Patch holography using a double layer microphone array

DEFF Research Database (Denmark)

Gomes, Jesper Skovhus

a closed local element mesh that surrounds the microphone array, and with a part of the mesh coinciding with a patch, the entire source is not needed in the model. Since the array has two layers, sources/reflections behind the array are also allowed. The Equivalent Source Method (ESM) is another technique...... in which the sound field is represented by a set of monopoles placed inside the source. In this paper these monopoles are distributed so that they surround the array, and the reconstruction is compared with the IBEM-based approach. The comparisons are based on computer simulations with a planar double...... layer array and sources with different shapes....

Calibration of High Frequency MEMS Microphones

Science.gov (United States)

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

2007-01-01

Understanding and controlling aircraft noise is one of the major research topics of the NASA Fundamental Aeronautics Program. One of the measurement technologies used to acquire noise data is the microphone directional array (DA). Traditional direction array hardware, consisting of commercially available condenser microphones and preamplifiers can be too expensive and their installation in hard-walled wind tunnel test sections too complicated. An emerging micro-machining technology coupled with the latest cutting edge technologies for smaller and faster systems have opened the way for development of MEMS microphones. The MEMS microphone devices are available in the market but suffer from certain important shortcomings. Based on early experiments with array prototypes, it has been found that both the bandwidth and the sound pressure level dynamic range of the microphones should be increased significantly to improve the performance and flexibility of the overall array. Thus, in collaboration with an outside MEMS design vendor, NASA Langley modified commercially available MEMS microphone as shown in Figure 1 to meet the new requirements. Coupled with the design of the enhanced MEMS microphones was the development of a new calibration method for simultaneously obtaining the sensitivity and phase response of the devices over their entire broadband frequency range. Over the years, several methods have been used for microphone calibration. Some of the common methods of microphone calibration are Coupler (Reciprocity, Substitution, and Simultaneous), Pistonphone, Electrostatic actuator, and Free-field calibration (Reciprocity, Substitution, and Simultaneous). Traditionally, electrostatic actuators (EA) have been used to characterize air-condenser microphones for wideband frequency ranges; however, MEMS microphones are not adaptable to the EA method due to their construction and very small diaphragm size. Hence a substitution-based, free-field method was developed to

Micromachined microphone array on a chip for turbulent boundary layer measurements

Science.gov (United States)

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

Beamforming with a circular microphone array for localization of environmental noise sources

DEFF Research Database (Denmark)

Tiana Roig, Elisabet; Jacobsen, Finn; Fernandez Grande, Efren

2010-01-01

It is often enough to localize environmental sources of noise from different directions in a plane. This can be accomplished with a circular microphone array, which can be designed to have practically the same resolution over 360. The microphones can be suspended in free space or they can...

Reconstruction of sound fields with a spherical microphone array

DEFF Research Database (Denmark)

Fernandez Grande, Efren; Walton, Tim

2014-01-01

waves traveling in any direction. In particular, rigid sphere microphone arrays are robust, and have the favorable property that the scattering introduced by the array can be compensated for - making the array virtually transparent. This study examines a recently proposed sound field reconstruction...... method based on a point source expansion, i.e. equivalent source method, using a rigid spherical array. The study examines the capability of the method to distinguish between sound waves arriving from different directions (i.e., as a sound field separation method). This is representative of the potential...

Methods for Room Acoustic Analysis and Synthesis using a Monopole-Dipole Microphone Array

Science.gov (United States)

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.

Improving beamforming by optimization of acoustic array microphone positions

NARCIS (Netherlands)

Malgoezar, A.M.N.; Snellen, M.; Sijtsma, P.; Simons, D.G.

2016-01-01

Assigning proper positions to microphones within arrays is essential in order to reduce or eliminate side- and grating lobes in 2D beamform images. In this paper an objective function is derived providing a measure for the presence of artificial sources. Using the global optimization method

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

Science.gov (United States)

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

2015-01-01

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

Motorcycle detection and counting using stereo camera, IR camera, and microphone array

Science.gov (United States)

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.

Sound-field reconstruction performance of a mixed-order Ambisonics microphone array

DEFF Research Database (Denmark)

Marschall, Marton; Chang, Jiho

2013-01-01

instruments and mobile phones. Previously, a mixed-order Ambisonics (MOA) approach was proposed to improve the horizontal spatial resolution of spherical arrays. This was achieved by increasing the number of microphones near the horizontal plane while keeping the total number of transducers fixed...

Phase Calibration of Microphones by Measurement in the Free-field

Science.gov (United States)

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

Fast calculation of microphone array steering vectors with shear flow

NARCIS (Netherlands)

Sijtsma, P.

2018-01-01

This paper proposes a fast method for calculating the acoustic time delay between an observer and a receiver in a shear flow. This method is applied to an outdoor microphone array measurement on a large-scale wind turbine. In such a set-up, a shear flow represents the actual wind field better than a

A Partitioned Approach to Signal Separation with Microphone Ad Hoc Arrays

DEFF Research Database (Denmark)

Tavakoli, Vincent Mohammad; Jensen, Jesper Rindom; Benesty, Jacob

2016-01-01

In this paper, a blind algorithm is proposed for speech enhancement in multi-speaker scenarios, in which interference rejection is the main objective. Here, the ad hoc array is broken into microphone duples which are used to partition the array into local sub-arrays. The core algorithm takes...... advantage of differences in signal structure in each duple. A geometric mean filter is then used to merge the output signals obtained with different duples, and to form a global broadband maximum signal-to-interference ratio (SIR) enhancement apparatus. The resulting filter outputs are enhanced acoustic...

Directional hearing aid using hybrid adaptive beamformer (HAB) and binaural ITE array

Science.gov (United States)

Shaw, Scott T.; Larow, Andy J.; Gibian, Gary L.; Sherlock, Laguinn P.; Schulein, Robert

2002-05-01

A directional hearing aid algorithm called the Hybrid Adaptive Beamformer (HAB), developed for NIH/NIA, can be applied to many different microphone array configurations. In this project the HAB algorithm was applied to a new array employing in-the-ear microphones at each ear (HAB-ITE), to see if previous HAB performance could be achieved with a more cosmetically acceptable package. With diotic output, the average benefit in threshold SNR was 10.9 dB for three HoH and 11.7 dB for five normal-hearing subjects. These results are slightly better than previous results of equivalent tests with a 3-in. array. With an innovative binaural fitting, a small benefit beyond that provided by diotic adaptive beamforming was observed: 12.5 dB for HoH and 13.3 dB for normal-hearing subjects, a 1.6 dB improvement over the diotic presentation. Subjectively, the binaural fitting preserved binaural hearing abilities, giving the user a sense of space, and providing left-right localization. Thus the goal of creating an adaptive beamformer that simultaneously provides excellent noise reduction and binaural hearing was achieved. Further work remains before the HAB-ITE can be incorporated into a real product, optimizing binaural adaptive beamforming, and integrating the concept with other technologies to produce a viable product prototype. [Work supported by NIH/NIDCD.

Estimation of surface impedance using different types of microphone arrays

DEFF Research Database (Denmark)

Richard, Antoine Philippe André; Fernandez Grande, Efren; Brunskog, Jonas

2017-01-01

This study investigates microphone array methods to measure the angle dependent surface impedance of acoustic materials. The methods are based on the reconstruction of the sound field on the surface of the material, using a wave expansion formulation. The reconstruction of both the pressure...... and the particle velocity leads to an estimation of the surface impedance for a given angle of incidence. A porous type absorber sample is tested experimentally in anechoic conditions for different array geometries, sample sizes, incidence angles, and distances between the array and sample. In particular......, the performances of a rigid spherical array and a double layer planar array are examined. The use of sparse array processing methods and conventional regulariation approaches are studied. In addition, the influence of the size of the sample on the surface impedance estimation is investigated using both...

Metrics for performance assessment of mixed-order Ambisonics spherical microphone arrays

DEFF Research Database (Denmark)

Favrot, Sylvain Emmanuel; Marschall, Marton

2012-01-01

Mixed-order Ambisonics (MOA) combines planar (2D) higher order Ambisonics (HOA) with lower order periphonic (3D) Ambisonics. MOA encoding from spherical microphone arrays has the potential to provide versatile recordings that can be played back using 2D, 3D or mixed systems. A procedure to generate...

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

Science.gov (United States)

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

2014-01-01

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

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

Science.gov (United States)

Bush, Dane; Xiang, Ning

2015-07-01

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

Use of a Microphone Phased Array to Determine Noise Sources in a Rocket Plume

Science.gov (United States)

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

On the influence of microphone array geometry on HRTF-based Sound Source Localization

DEFF Research Database (Denmark)

Farmani, Mojtaba; Pedersen, Michael Syskind; Tan, Zheng-Hua

2015-01-01

The direction dependence of Head Related Transfer Functions (HRTFs) forms the basis for HRTF-based Sound Source Localization (SSL) algorithms. In this paper, we show how spectral similarities of the HRTFs of different directions in the horizontal plane influence performance of HRTF-based SSL...... algorithms; the more similar the HRTFs of different angles to the HRTF of the target angle, the worse the performance. However, we also show how the microphone array geometry can assist in differentiating between the HRTFs of the different angles, thereby improving performance of HRTF-based SSL algorithms....... Furthermore, to demonstrate the analysis results, we show the impact of HRTFs similarities and microphone array geometry on an exemplary HRTF-based SSL algorithm, called MLSSL. This algorithm is well-suited for this purpose as it allows to estimate the Direction-of-Arrival (DoA) of the target sound using any...

Design and preliminary testing of a MEMS microphone phased array for aeroacoustic testing of a small-scale wind turbine airfoil

Energy Technology Data Exchange (ETDEWEB)

Bale, A.; Orlando, S.; Johnson, D. [Waterloo Univ., ON (Canada). Wind Energy Group

2010-07-01

One of the barriers preventing the widespread utilization of wind turbines is the audible sound that they produce. Developing quieter wind turbines will increase the amount of available land onto which wind farms can be built. Noise emissions from wind turbines can be attributed to the aerodynamic effects between the turbine blades and the air surrounding them. A dominant source of these aeroacoustic emissions from wind turbines is known to originate at the trailing edges of the airfoils. This study investigated the flow physics of noise generation in an effort to reduce noise from small-scale wind turbine airfoils. The trailing edge noise was studied on scale-models in wind tunnels and applied to full scale conditions. Microphone phased arrays are popular research tools in wind tunnel aeroacoustic studies because they can measure and locate noise sources. However, large arrays of microphones can be prohibitively expensive. This paper presented preliminary testing of micro-electrical mechanical system (MEMS) microphones in phased arrays for aeroacoustic testing on a small wind turbine airfoil. Preliminary results showed that MEMS microphones are an acceptable low-cost alternative to costly condenser microphones. 19 refs., 1 tab., 11 figs.

Investigation of noise radiation from a swirl stabilized diffusion flame with an array of microphones

International Nuclear Information System (INIS)

Singh, A.V.; Yu, M.; Gupta, A.K.; Bryden, K.M.

2013-01-01

Highlights: • Acoustic spectral characteristics independent of equivalence ratio and flow velocity. • Combustion noise dependent on global equivalence ratio and flow velocity. • Increased global equivalence ratio decreased the frequency of peak. • Decay and growth coefficients largely independent of different flow conditions. • Acoustic radiation coherent up to 1.5 kHz for spatially separated microphones. - Abstract: Next generation of combustors are expected to provide significant improvement on efficiency and reduced pollutants emission. In such combustors, the challenges of local flow, pressure, chemical composition and thermal signatures as well as their interactions will require detailed investigation for seeking optimum performance. Sensor networks with a large number of sensors will be employed in future smart combustors, which will allow one to obtain fast and comprehensive information on the various ongoing processes within the system. In this paper sensor networks with specific focus on an array of homogeneous microphones are used examine the spectral characteristics of combustion noise from a non-premixed combustor. A non-premixed double concentric swirl-flame burner was used. Noise spectra were determined experimentally for the non-premixed swirl flame at various fuel–air ratios using an array of homogeneous condenser microphones. Multiple microphones positioned at discrete locations around the turbulent diffusion flame, provided an understanding of the total sound power and their spectral characteristics. The growth and decay coefficients of total sound power were investigated at different test conditions. The signal coherence between different microphone pairs was also carried out to determine the acoustic behavior of a swirl stabilized turbulent diffusion flame. The localization of acoustic sources from the multiple microphones was examined using the noise spectra. The results revealed that integration of multiple sensors in combustors

Practically Efficient Blind Speech Separation Using Frequency Band Selection Based on Magnitude Squared Coherence and a Small Dodecahedral Microphone Array

Directory of Open Access Journals (Sweden)

Kazunobu Kondo

2012-01-01

Full Text Available Small agglomerative microphone array systems have been proposed for use with speech communication and recognition systems. Blind source separation methods based on frequency domain independent component analysis have shown significant separation performance, and the microphone arrays are small enough to make them portable. However, the level of computational complexity involved is very high because the conventional signal collection and processing method uses 60 microphones. In this paper, we propose a band selection method based on magnitude squared coherence. Frequency bands are selected based on the spatial and geometric characteristics of the microphone array device which is strongly related to the dodecahedral shape, and the selected bands are nonuniformly spaced. The estimated reduction in the computational complexity is 90% with a 68% reduction in the number of frequency bands. Separation performance achieved during our experimental evaluation was 7.45 (dB (signal-to-noise ratio and 2.30 (dB (cepstral distortion. These results show improvement in performance compared to the use of uniformly spaced frequency band.

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

Science.gov (United States)

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

2011-08-01

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

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

Science.gov (United States)

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

2018-03-01

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

Studying Room Acoustics using a Monopole-Dipole Microphone Array

Science.gov (United States)

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

1997-01-01

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

Pseudo-Coherence-Based MVDR Beamformer for Speech Enhancement with Ad Hoc Microphone Arrays

DEFF Research Database (Denmark)

Tavakoli, Vincent Mohammad; Jensen, Jesper Rindom; Christensen, Mads Græsbøll

2015-01-01

Speech enhancement with distributed arrays has been met with various methods. On the one hand, data independent methods require information about the position of sensors, so they are not suitable for dynamic geometries. On the other hand, Wiener-based methods cannot assure a distortionless output....... This paper proposes minimum variance distortionless response filtering based on multichannel pseudo-coherence for speech enhancement with ad hoc microphone arrays. This method requires neither position information nor control of the trade-off used in the distortion weighted methods. Furthermore, certain...

Periphony-Lattice Mixed-Order Ambisonic Scheme for Spherical Microphone Arrays

DEFF Research Database (Denmark)

Chang, Jiho; Marschall, Marton

2018-01-01

to performance that is independent of the incident direction of the sound waves. On the other hand, mixed-order ambisonic (MOA) schemes that select an appropriate subset of spherical harmonics can improve the performance for horizontal directions at the expense of other directions. This paper proposes an MOA......Most methods for sound field reconstruction and spherical beamforming with spherical microphone arrays are mathematically based on the spherical harmonics expansion. In many cases, this expansion is truncated at a certain order as in higher order ambisonics (HOA). This truncation leads...

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

Science.gov (United States)

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.

Ad Hoc Microphone Array Beamforming Using the Primal-Dual Method of Multipliers

DEFF Research Database (Denmark)

Tavakoli, Vincent Mohammad; Jensen, Jesper Rindom; Heusdens, Richard

2016-01-01

In the recent years, there have been increasing amount of researches aiming at optimal beamforming with ad hoc microphone arrays, mostly with fusion-based schemes. However, huge amount of computational complexity and communication overhead impede many of these algorithms from being useful in prac...... the distributed linearly-constrained minimum variance beamformer using the the state of the art primal-dual method of multipliers. We study the proposed algorithm with an experiment....

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

Science.gov (United States)

Fleury, Vincent; Davy, Renaud

2016-03-01

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

Development of the microphone array measurement technique for application to cryogenic wind tunnels; Entwicklung der Mikrofonarraymesstechnik fuer die experimentelle Anwendung in kryogenen Windkanaelen

Energy Technology Data Exchange (ETDEWEB)

Ahlefeldt, Thomas

2013-02-01

The present work deals with the development of the microphone array measurement technique for application to cryogenic wind tunnels at temperatures down to 100 K. In contrast to conventional wind tunnels, in cryogenic wind tunnels the Reynolds number can be changed independent of the Mach number. Therefore the applicability of the microphone array measurement technique to cryogenic wind tunnels allows the independent investigation of Mach and Reynolds number effects for aeroacoustic sources. For this purpose two microphone arrays suitable for cryogenic application have been developed. A small array was used for a validation experiment using a single-rod configuration as an aeroacoustic noise source; the experience gained therefrom being then used to develop a larger array. This array was used to finally demonstrate the applicability of the measuring technology to an airplane half model. For the development of both arrays several factors had to be considered, such as, for example, the contraction arising from the low temperatures and the influence of the temperature on the microphone frequency response. In the validation experiment, acoustic array measurements have been performed using the small microphone array with 21 microphones in a cryogenic wind tunnel for various Mach and Reynolds numbers, using a single-rod configuration. The aeroacoustic source induced by the rod could be identified by the microphone.array at ambient as well as at cryogenic temperatures. The radiated sound powers were compared with predictions from two models: one model was based on a dimensional analysis of the measured data without taking into consideration the Reynolds number. The measured data with this model could be better fitted by a speed law with the exponent 6.7 rather than the expected 6.0. The second model was based on an analytical model for sound radiation from a single-rod configuration which took into account variables dependent on the Reynolds number. The comparison with

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

International Nuclear Information System (INIS)

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

2017-01-01

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

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

Science.gov (United States)

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

2017-10-01

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

Application of a circular 2D hard-sphere microphone array for higher-order Ambisonics auralization

DEFF Research Database (Denmark)

Weller, Tobias; Favrot, Sylvain Emmanuel; Buchholz, Jörg

2011-01-01

. The simulation results showed very good agreement with corresponding plane wave recordings in an anechoic chamber and thus, confirming the general applicability of the simulation framework. An overall preference listening test was performed to estimate the optimal array radius and amount of regularization, two...... (dependent) parameters that mainly determine the balance between low frequency directionality, signal coloration and microphone noise amplification. The different stimuli were created with the framework using different values for both the array radius and the regularization coefficient lambda. It was shown...

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

Science.gov (United States)

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

2016-01-01

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

Plane-wave decomposition by spherical-convolution microphone array

Science.gov (United States)

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.

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

Science.gov (United States)

Moghaddasi, Hanie; Almasganj, Farshad; Zoroufian, Arezoo

2017-07-01

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

Compressing Sensing Based Source Localization for Controlled Acoustic Signals Using Distributed Microphone Arrays

Directory of Open Access Journals (Sweden)

Wei Ke

2017-01-01

Full Text Available In order to enhance the accuracy of sound source localization in noisy and reverberant environments, this paper proposes an adaptive sound source localization method based on distributed microphone arrays. Since sound sources lie at a few points in the discrete spatial domain, our method can exploit this inherent sparsity to convert the localization problem into a sparse recovery problem based on the compressive sensing (CS theory. In this method, a two-step discrete cosine transform- (DCT- based feature extraction approach is utilized to cover both short-time and long-time properties of acoustic signals and reduce the dimensions of the sparse model. In addition, an online dictionary learning (DL method is used to adjust the dictionary for matching the changes of audio signals, and then the sparse solution could better represent location estimations. Moreover, we propose an improved block-sparse reconstruction algorithm using approximate l0 norm minimization to enhance reconstruction performance for sparse signals in low signal-noise ratio (SNR conditions. The effectiveness of the proposed scheme is demonstrated by simulation results and experimental results where substantial improvement for localization performance can be obtained in the noisy and reverberant conditions.

Active microphonic noise cancellation in radiation detectors

International Nuclear Information System (INIS)

Zimmermann, Sergio

2013-01-01

A new adaptive filtering technique to reduce microphonic noise in radiation detectors is presented. The technique is based on system identification that actively cancels the microphonic noise. A sensor is used to measures mechanical disturbances that cause vibration on the detector assembly, and the digital adaptive filtering estimates the impact of these disturbances on the microphonic noise. The noise then can be subtracted from the actual detector measurement. In this paper the technique is presented and simulations are used to support this approach. -- Highlights: •A sensor is used to measures mechanical disturbances that cause vibration on the detector assembly. •Digital adaptive filtering estimates the impact of these disturbances on the microphonic noise. •The noise is then subtracted from the actual detector measurement. •We use simulations to demonstrate the performance of this approach. •After cancellation, we recover most of the original energy resolution

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

Science.gov (United States)

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.

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

DEFF Research Database (Denmark)

Tiana Roig, Elisabet; Jacobsen, Finn

2013-01-01

During the last decade, the aeroacoustic community has examined various methods based on deconvolution to improve the visualization of acoustic fields scanned with planar sparse arrays of microphones. These methods assume that the beamforming map in an observation plane can be approximated by a c......-negative least squares, and the Richardson-Lucy. This investigation examines the matter with computer simulations and measurements....... that the beamformer's point-spread function is shift-invariant. This makes it possible to apply computationally efficient deconvolution algorithms that consist of spectral procedures in the entire region of interest, such as the deconvolution approach for the mapping of the acoustic sources 2, the Fourier-based non...

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

Science.gov (United States)

Garcia, Roberto

2012-01-01

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

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

Science.gov (United States)

Chen, Yung-Yue

2018-05-08

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

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

Directory of Open Access Journals (Sweden)

Yung-Yue Chen

2018-05-01

Full Text Available Mobile devices are often used in our daily lives for the purposes of speech and communication. The speech quality of mobile devices is always degraded due to the environmental noises surrounding mobile device users. Regretfully, an effective background noise reduction solution cannot easily be developed for this speech enhancement problem. Due to these depicted reasons, a methodology is systematically proposed to eliminate the effects of background noises for the speech communication of mobile devices. This methodology integrates a dual microphone array with a background noise elimination algorithm. The proposed background noise elimination algorithm includes a whitening process, a speech modelling method and an H2 estimator. Due to the adoption of the dual microphone array, a low-cost design can be obtained for the speech enhancement of mobile devices. Practical tests have proven that this proposed method is immune to random background noises, and noiseless speech can be obtained after executing this denoise process.

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

Science.gov (United States)

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

2007-02-01

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

Wheel/rail noise generated by a high-speed train investigated with a line array of microphones

Science.gov (United States)

Barsikow, B.; King, W. F.; Pfizenmaier, E.

1987-10-01

Radiated noise generated by a high-speed electric train travelling at speeds up to 250 km/h has been measured with a line array of microphones mounted along the wayside in two different orientations. The test train comprised a 103 electric locomotive, four Intercity coaches, and a dynamo coach. Some of the wheels were fitted with experimental wheel-noise absorbers. By using the directional capabilities of the array, the locations of the dominant sources of wheel/rail radiated noise were identified on the wheels. For conventional wheels, these sources lie near or on the rim at an average height of about 0·2 m above the railhead. The effect of wheel-noise absorbers and freshly turned treads on radiated noise were also investigated.

Improving source discrimination performance by using an optimized acoustic array and adaptive high-resolution CLEAN-SC beamforming

NARCIS (Netherlands)

Luesutthiviboon, S.; Malgoezar, A.M.N.; Snellen, M.; Sijtsma, P.; Simons, D.G.

2018-01-01

Beamforming performance can be improved in two ways: optimizing the location of microphones on the acoustic array and applying advanced beamforming algorithms. In this study, the effects of the two approaches are studied. An optimization method is developed to optimize the location of microphones

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

Science.gov (United States)

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

2017-07-01

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

Source Coding for Wireless Distributed Microphones in Reverberant Environments

DEFF Research Database (Denmark)

Zahedi, Adel

2016-01-01

. However, it comes with the price of several challenges, including the limited power and bandwidth resources for wireless transmission of audio recordings. In such a setup, we study the problem of source coding for the compression of the audio recordings before the transmission in order to reduce the power...... consumption and/or transmission bandwidth by reduction in the transmission rates. Source coding for wireless microphones in reverberant environments has several special characteristics which make it more challenging in comparison with regular audio coding. The signals which are acquired by the microphones......Modern multimedia systems are more and more shifting toward distributed and networked structures. This includes audio systems, where networks of wireless distributed microphones are replacing the traditional microphone arrays. This allows for flexibility of placement and high spatial diversity...

Introduction to adaptive arrays

CERN Document Server

Monzingo, Bob; Haupt, Randy

2011-01-01

This second edition is an extensive modernization of the bestselling introduction to the subject of adaptive array sensor systems. With the number of applications of adaptive array sensor systems growing each year, this look at the principles and fundamental techniques that are critical to these systems is more important than ever before. Introduction to Adaptive Arrays, 2nd Edition is organized as a tutorial, taking the reader by the hand and leading them through the maze of jargon that often surrounds this highly technical subject. It is easy to read and easy to follow as fundamental concept

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

Science.gov (United States)

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.

Spherical near field acoustic holography with microphones on a rigid sphere

DEFF Research Database (Denmark)

Jacobsen, Finn; Hald, Jørgen; Fernandez Grande, Efren

2008-01-01

Spherical near field acoustic holography (SNAH) is a recently developed technique that makes it possible to reconstruct the sound field inside and just outside an acoustically transparent spherical surface on which the sound pressure is measured with an array of microphones with negligible...... with an array of microphones flush-mounted on a rigid sphere. However, this approach is only valid if it can be assumed that the sphere has a negligible influence on the incident sound field, in other words if multiple scattering can be ignored, and this is not necessarily a good assumption when the sphere...

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

Science.gov (United States)

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

2017-10-13

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

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

Science.gov (United States)

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

Numerical Analysis of CNC Milling Chatter Using Embedded Miniature MEMS Microphone Array System

Directory of Open Access Journals (Sweden)

Pang-Li Wang

2018-01-01

Full Text Available With the increasingly common use of industrial automation for mass production, there are many computer numerical control (CNC machine tools that require the collection of data from intelligent sensors in order to analyze their processing quality. In general, for high speed rotating machines, an accelerometer can be attached on the spindle to collect the data from the detected vibration of the CNC. However, due to their cost, accelerometers have not been widely adopted for use with typical CNC machine tools. This study sought to develop an embedded miniature MEMS microphone array system (Radius 5.25 cm, 8 channels to discover the vibration source of the CNC from spatial phase array processing. The proposed method utilizes voice activity detection (VAD to distinguish between the presence and absence of abnormal noise in the pre-stage, and utilizes the traditional direction of arrival method (DOA via multiple signal classification (MUSIC to isolate the spatial orientation of the noise source in post-processing. In the numerical simulation, the non-interfering noise source location is calibrated in the anechoic chamber, and is tested with real milling processing in the milling machine. As this results in a high background noise level, the vibration sound source is more accurate in the presented energy gradation graphs as compared to the traditional MUSIC method.

Acoustic array systems theory, implementation, and application

CERN Document Server

Bai, Mingsian R; Benesty, Jacob

2013-01-01

Presents a unified framework of far-field and near-field array techniques for noise source identification and sound field visualization, from theory to application. Acoustic Array Systems: Theory, Implementation, and Application provides an overview of microphone array technology with applications in noise source identification and sound field visualization. In the comprehensive treatment of microphone arrays, the topics covered include an introduction to the theory, far-field and near-field array signal processing algorithms, practical implementations, and common applic

Microphone variability and degradation: implications for monitoring programs employing autonomous recording units

Directory of Open Access Journals (Sweden)

Patrick J. Turgeon

2017-06-01

Full Text Available Autonomous recording units (ARUs are emerging as an effective tool for avian population monitoring and research. Although ARU technology is being rapidly adopted, there is a need to establish whether variation in ARU components and their degradation with use might introduce detection biases that would affect long-term monitoring and research projects. We assessed whether microphone sensitivity impacted the probability of detecting bird vocalizations by broadcasting a sequence of 12 calls toward an array of commercially available ARUs equipped with microphones of varying sensitivities under three levels (32 dBA, 42 dBA, and 50 dBA of experimentally induced noise conditions selected to reflect the range of noise levels commonly encountered during avian surveys. We used binomial regression to examine factors influencing probability of detection for each species and used these to examine the impact of microphone sensitivity on the effective detection area (ha for each species. Microphone sensitivity loss reduced detection probability for all species examined, but the magnitude of the effect varied between species and often interacted with distance. Microphone sensitivity loss reduced the effective detection area by an average of 25% for microphones just beyond manufacturer specifications (-5 dBV and by an average of 66% for severely compromised microphones (-20 dBV. Microphone sensitivity loss appeared to be more problematic for low frequency calls where reduction in the effective detection area occurred most rapidly. Microphone degradation poses a source of variation in avian surveys made with ARUs that will require regular measurement of microphone sensitivity and criteria for microphone replacement to ensure scientifically reproducible results. We recommend that research and monitoring projects employing ARUs test their microphones regularly, replace microphones with declining sensitivity, and record sensitivity as a potential covariate in

Robust Nearfield Wideband Beamforming Design Based on Adaptive-Weighted Convex Optimization

Directory of Open Access Journals (Sweden)

Guo Ye-Cai

2017-01-01

Full Text Available Nearfield wideband beamformers for microphone arrays have wide applications in multichannel speech enhancement. The nearfield wideband beamformer design based on convex optimization is one of the typical representatives of robust approaches. However, in this approach, the coefficient of convex optimization is a constant, which has not used all the freedom provided by the weighting coefficient efficiently. Therefore, it is still necessary to further improve the performance. To solve this problem, we developed a robust nearfield wideband beamformer design approach based on adaptive-weighted convex optimization. The proposed approach defines an adaptive-weighted function by the adaptive array signal processing theory and adjusts its value flexibly, which has improved the beamforming performance. During each process of the adaptive updating of the weighting function, the convex optimization problem can be formulated as a SOCP (Second-Order Cone Program problem, which could be solved efficiently using the well-established interior-point methods. This method is suitable for the case where the sound source is in the nearfield range, can work well in the presence of microphone mismatches, and is applicable to arbitrary array geometries. Several design examples are presented to verify the effectiveness of the proposed approach and the correctness of the theoretical analysis.

Fundamentals of spherical array processing

CERN Document Server

Rafaely, Boaz

2015-01-01

This book provides a comprehensive introduction to the theory and practice of spherical microphone arrays. It is written for graduate students, researchers and engineers who work with spherical microphone arrays in a wide range of applications.   The first two chapters provide the reader with the necessary mathematical and physical background, including an introduction to the spherical Fourier transform and the formulation of plane-wave sound fields in the spherical harmonic domain. The third chapter covers the theory of spatial sampling, employed when selecting the positions of microphones to sample sound pressure functions in space. Subsequent chapters present various spherical array configurations, including the popular rigid-sphere-based configuration. Beamforming (spatial filtering) in the spherical harmonics domain, including axis-symmetric beamforming, and the performance measures of directivity index and white noise gain are introduced, and a range of optimal beamformers for spherical arrays, includi...

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

Science.gov (United States)

Mens, Lucas H M

2011-01-01

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

Near field acoustic holography with microphones on a rigid sphere

DEFF Research Database (Denmark)

Jacobsen, Finn; Moreno-Pescador, Guillermo; Fernandez Grande, Efren

2011-01-01

Spherical near field acoustic holography (spherical NAH) is a technique that makes it possible to reconstruct the sound field inside and just outside a spherical surface on which the sound pressure is measured with an array of microphones. This is potentially very useful for source identification...

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

NARCIS (Netherlands)

Mens, L.H.M.

2011-01-01

OBJECTIVE: To test speech understanding in noise using array microphones integrated in an eyeglass device and to test if microphones placed anteriorly at the temple provide better directivity than above the pinna. DESIGN: Sentences were presented from the front and uncorrelated noise from 45, 135,

High Channel Count, High Density Microphone Arrays for Wind Tunnel Environments, Phase I

Data.gov (United States)

National Aeronautics and Space Administration — The Interdisciplinary Consulting Corporation (IC2) proposes the development of high channel count, high density, reduced cost per channel, directional microphone...

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

Science.gov (United States)

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.

Mic it! microphones, microphone techniques, and their impact on the final mix

CERN Document Server

Corbett, Ian

2014-01-01

Capture great sound in the first place, and spend less time ""fixing it in the mix"" with Ian Corbett's Mic It! Microphones, Microphone Techniques, and Their Impact on the Final Mix. With his expert guidance, you'll quickly understand essential audio concepts as they relate to microphones and mic techniques, and learn how to apply them to your recording situation. Whether you only ever buy one microphone, are equipping a studio on a budget, or have a vast selection of great mics to use, you'll learn to better use whatever tools you have. Mic It! gives you the background to design and discover

Mapping Speech Spectra from Throat Microphone to Close-Speaking Microphone: A Neural Network Approach

Directory of Open Access Journals (Sweden)

B. Yegnanarayana

2007-01-01

Full Text Available Speech recorded from a throat microphone is robust to the surrounding noise, but sounds unnatural unlike the speech recorded from a close-speaking microphone. This paper addresses the issue of improving the perceptual quality of the throat microphone speech by mapping the speech spectra from the throat microphone to the close-speaking microphone. A neural network model is used to capture the speaker-dependent functional relationship between the feature vectors (cepstral coefficients of the two speech signals. A method is proposed to ensure the stability of the all-pole synthesis filter. Objective evaluations indicate the effectiveness of the proposed mapping scheme. The advantage of this method is that the model gives a smooth estimate of the spectra of the close-speaking microphone speech. No distortions are perceived in the reconstructed speech. This mapping technique is also used for bandwidth extension of telephone speech.

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

Science.gov (United States)

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

Removing Background Noise with Phased Array Signal Processing

Science.gov (United States)

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.

Locating and Quantifying Broadband Fan Sources Using In-Duct Microphones

Science.gov (United States)

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.

Two-wire Interface for Digital Microphones

NARCIS (Netherlands)

Groothedde, Wouter; Klumperink, Eric A.M.; Nauta, Bram; Eschauzier, Rudolphe Gustave Hubertus; van Rijn, Nico

2003-01-01

A two-wire interface for a digital microphone circuit includes a power line and a ground line. The interface utilizes the ground line as a "voltage active line" to transmit both clock and data signals between the digital microphone circuit and a receiving circuit. The digital microphone circuit

Two-Wire interface for digital microphones

NARCIS (Netherlands)

Groothedde, Wouter; Klumperink, Eric A.M.; Nauta, Bram; Eschauzier, Rudolphe Gustave Hubertus; van Rijn, Nico

2005-01-01

A two-wire interface for a digital microphone circuit includes a power line and a ground line. The interface utilizes the ground line as a "voltage active line" to transmit both clock and data signals between the digital microphone circuit and a receiving circuit. The digital microphone circuit

FPGA implementation of adaptive beamforming in hearing aids.

Science.gov (United States)

Samtani, Kartik; Thomas, Jobin; Varma, G Abhinav; Sumam, David S; Deepu, S P

2017-07-01

Beamforming is a spatial filtering technique used in hearing aids to improve target sound reception by reducing interference from other directions. In this paper we propose improvements in an existing architecture present for two omnidirectional microphone array based adaptive beamforming for hearing aid applications and implement the same on Xilinx Artix 7 FPGA using VHDL coding and Xilinx Vivado ® 2015.2. The nulls are introduced in particular directions by combination of two fixed polar patterns. This combination can be adaptively controlled to steer the null in the direction of noise. The beamform patterns and improvements in SNR values obtained from experiments in a conference room environment are analyzed.

Sparse acoustic imaging with a spherical array

DEFF Research Database (Denmark)

Fernandez Grande, Efren; Xenaki, Angeliki

2015-01-01

In recent years, a number of methods for sound source localization and sound field reconstruction with spherical microphone arrays have been proposed. These arrays have properties that are potentially very useful, e.g. omni-directionality, robustness, compensable scattering, etc. This paper propo...

Acoustic Source Localization via Subspace Based Method Using Small Aperture MEMS Arrays

Directory of Open Access Journals (Sweden)

Xin Zhang

2014-01-01

Full Text Available Small aperture microphone arrays provide many advantages for portable devices and hearing aid equipment. In this paper, a subspace based localization method is proposed for acoustic source using small aperture arrays. The effects of array aperture on localization are analyzed by using array response (array manifold. Besides array aperture, the frequency of acoustic source and the variance of signal power are simulated to demonstrate how to optimize localization performance, which is carried out by introducing frequency error with the proposed method. The proposed method for 5 mm array aperture is validated by simulations and experiments with MEMS microphone arrays. Different types of acoustic sources can be localized with the highest precision of 6 degrees even in the presence of wind noise and other noises. Furthermore, the proposed method reduces the computational complexity compared with other methods.

Evaluation of Speech Recognition of Cochlear Implant Recipients Using Adaptive, Digital Remote Microphone Technology and a Speech Enhancement Sound Processing Algorithm.

Science.gov (United States)

Wolfe, Jace; Morais, Mila; Schafer, Erin; Agrawal, Smita; Koch, Dawn

2015-05-01

Cochlear implant recipients often experience difficulty with understanding speech in the presence of noise. Cochlear implant manufacturers have developed sound processing algorithms designed to improve speech recognition in noise, and research has shown these technologies to be effective. Remote microphone technology utilizing adaptive, digital wireless radio transmission has also been shown to provide significant improvement in speech recognition in noise. There are no studies examining the potential improvement in speech recognition in noise when these two technologies are used simultaneously. The goal of this study was to evaluate the potential benefits and limitations associated with the simultaneous use of a sound processing algorithm designed to improve performance in noise (Advanced Bionics ClearVoice) and a remote microphone system that incorporates adaptive, digital wireless radio transmission (Phonak Roger). A two-by-two way repeated measures design was used to examine performance differences obtained without these technologies compared to the use of each technology separately as well as the simultaneous use of both technologies. Eleven Advanced Bionics (AB) cochlear implant recipients, ages 11 to 68 yr. AzBio sentence recognition was measured in quiet and in the presence of classroom noise ranging in level from 50 to 80 dBA in 5-dB steps. Performance was evaluated in four conditions: (1) No ClearVoice and no Roger, (2) ClearVoice enabled without the use of Roger, (3) ClearVoice disabled with Roger enabled, and (4) simultaneous use of ClearVoice and Roger. Speech recognition in quiet was better than speech recognition in noise for all conditions. Use of ClearVoice and Roger each provided significant improvement in speech recognition in noise. The best performance in noise was obtained with the simultaneous use of ClearVoice and Roger. ClearVoice and Roger technology each improves speech recognition in noise, particularly when used at the same time

The acoustic center of laboratory standard microphones

DEFF Research Database (Denmark)

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

2006-01-01

An experimental procedure is described for obtaining the effective acoustic distance between pairs of microphones coupled by a free field, leading to the determination of the position of the acoustic center of the microphones. The procedure, which is based on measuring the modulus of the electrical...... transfer impedance, has been applied to a large number of microphones. In all cases effects due to reflections from the walls of the anechoic chamber and the interference between the microphones have been removed using a time-selective technique. The procedure of determining the position of the acoustic...... center from the inverse distance law is analyzed. Experimental values of the acoustic center of laboratory standard microphones are presented, and numerical results obtained using the boundary element method supplement the experimental data. Estimated uncertainties are also presented. The results...

A chain of microphones

International Nuclear Information System (INIS)

1994-07-01

In order to discover a more accurate and selective measuring method for the identification of individual flow-noise pollution sources on wind turbines blades, measuring equipment based on a chain of microphones was developed. The principle underlying the design of this equipment is that signals from a number of microphones can be interpreted. Thus the microphones can register noise from sections of the rotary blade and unwished-for noise is eliminated. The gating technique ensures that noises from individual blades can be separated and that clarity is improved. In addition to this, noise can be determined close to the source. The chain consists of 8 microphones placed in a row at adjustable distances. Measurements are registered on tapes as are the trigger signals for the blade passage. The computer processes the measurement results and unnecessary noise is depressed. The listening angles can also be changed electronically so that the doppler effect can be corrected. Results confirmed that the equipment operated satisfactorily and could also be used in relation to noise pollution in power plants as it is especially effective in depressing excess, and cutting out outside, noise and registers accurately individual sources of noise helped by its ability to ''listen '' at varying angles to the source. (AB)

Subband Adaptive Array for DS-CDMA Mobile Radio

Directory of Open Access Journals (Sweden)

Tran Xuan Nam

2004-01-01

Full Text Available We propose a novel scheme of subband adaptive array (SBAA for direct-sequence code division multiple access (DS-CDMA. The scheme exploits the spreading code and pilot signal as the reference signal to estimate the propagation channel. Moreover, instead of combining the array outputs at each output tap using a synthesis filter and then despreading them, we despread directly the array outputs at each output tap by the desired user's code to save the synthesis filter. Although its configuration is far different from that of 2D RAKEs, the proposed scheme exhibits relatively equivalent performance of 2D RAKEs while having less computation load due to utilising adaptive signal processing in subbands. Simulation programs are carried out to explore the performance of the scheme and compare its performance with that of the standard 2D RAKE.

Superconducting microphone for photoacoustic spectroscopy

International Nuclear Information System (INIS)

Ribeiro, P.C.; Labrunie, M.; Weid, J.P. von der; Symko, O.G.

1982-07-01

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 a SQUID magnetometer. For the simple set-up presented here, the limiting pressure sensitivity is 7.5x10 -14 atmospheres/√Hz. (Author) [pt

Speech Intelligibility in Noise Using Throat and Acoustic Microphones

National Research Council Canada - National Science Library

Acker-Mills, Barbara

2004-01-01

... speech intelligibility. Speech intelligibility for signals generated by an acoustic microphone, a throat microphone, and the two microphones together was assessed using the Modified Rhyme Test (MRT...

Implementation of Hybrid Speech Dereverberation Systems and Proposing Dual Microphone Farsi Database in Order to Evaluating Enhancement Systems

Directory of Open Access Journals (Sweden)

Farhad Faghani

2013-01-01

Full Text Available In various applications, such as speech recognition and automatic teleconferencing, the recorded speech signals may be corrupted by both noise and reverberation. Reverberation causes a noticeable change in speech intelligibility and quality. In this research, firstly reverberation is described. There are some de-reverberation enhancement algorithms that use only one microphone. They mostly use inverse filtering and spectral subtraction as their sub-systems. On the other hand, there are many multi-microphone speech enhancement systems; Delay-and-sum beam former is the most famous amongst them. Moreover, several efficient approaches have been also reported that use linear prediction (LP residual signal, inverse filtering, and phase error. Despite the improvements and benefits gained by the use of several input microphones, considering the tradeoff between these gains and the complexity and computational cost forced by the use of more microphones, many researchers have focused on dual-microphones systems. So, a review on Microphone array signal processing is explained and then an arrangement for two microphones systems is proposed. As we want to evaluate these algorithms for Farsi speech signals, the problem of speech intelligibility assessment has been explained and a Farsi word list for Diagnostic Rhyme Test (DRT is presented.The structure of presented word list is similar to that of English DRT words. In this research, after a brief study of above-mentioned methods, we propose and implement some hybrid techniques to benefit from the advantages of several methods and achieve significant improvement in output signals. It will be shown that the proposed method performs superior to the state-of-the-art dereverberation algorithms.

Silicon microphones - a Danish perspective

DEFF Research Database (Denmark)

Bouwstra, Siebe; Storgaard-Larsen, Torben; Scheeper, Patrick

1998-01-01

Two application areas of microphones are discussed, those for precision measurement and those for hearing instruments. Silicon microphones are under investigation for both areas, and Danish industry plays a key role in both. The opportunities of silicon, as well as the challenges and expectations......, are discussed. For precision measurement the challenge for silicon is large, while for hearing instruments silicon seems to be very promising....

Principles of Adaptive Array Processing

Science.gov (United States)

2006-09-01

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

Adaptive ground implemented phase array

Science.gov (United States)

Spearing, R. E.

1973-01-01

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

Optical microphone with fiber Bragg grating and signal processing techniques

Science.gov (United States)

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.

The ribbon microphone - an educational aid: use of a ribbon microphone to teach multi-discipline computer simulation skills

CSIR Research Space (South Africa)

Van Wyk, Marius

2016-07-01

Full Text Available The ribbon microphone serves as an excellent aid to learn computer simulation and computational skills. Simulation of this seemingly simple device is all but trivial. The ribbon microphone is an all-in-one example for simulations in acoustics...

Sodium immersible high temperature microphone design description

International Nuclear Information System (INIS)

Gavin, A.P.; Anderson, T.T.; Janicek, J.J.

1975-02-01

Argonne National Laboratory has developed a rugged high-temperature (HT) microphone for use as a sodium-immersed acoustic monitor in Liquid Metal Fast Breeder Reactors (LMFBRs). Microphones of this design have been extensively tested in room temperature water, in air up to 1200 0 F, and in sodium up to 1200 0 F. They have been successfully installed and employed as acoustic monitors in several operating liquid metal systems. The design, construction sequence, calibration, and testing of these microphones are described. 6 references. (U.S.)

MISSION-ORIENTED SENSOR ARRAYS AND UAVs – A CASE STUDY ON ENVIRONMENTAL MONITORING

Directory of Open Access Journals (Sweden)

N. M. Figueira

2015-08-01

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

Comparison of Multiple-Microphone and Single-Loudspeaker Adaptive Feedback/Echo Cancellation Systems

DEFF Research Database (Denmark)

Guo, Meng; Elmedyb, Thomas Bo; Jensen, Søren Holdt

2011-01-01

Recently, we introduced a frequency domain measure - the power transfer function - to predict the convergence rate, system stability bound and the steady-state behavior across time and frequency of a least mean square based feedback/echo cancellation algorithm in a general multiple-microphone and......Recently, we introduced a frequency domain measure - the power transfer function - to predict the convergence rate, system stability bound and the steady-state behavior across time and frequency of a least mean square based feedback/echo cancellation algorithm in a general multiple...

MICROMECHANICAL MICROPHONE

DEFF Research Database (Denmark)

1997-01-01

and dirt, which partly or totally will be able to destroy its characteristics, a sealing acoustic membrane (6, 7) is placed on each side of the transducer element. The transducer element can for example be a capacitive transducer with external bias or an electret based transducer. The microphone, which can...

Microphonic measurements on superconducting linac structures

International Nuclear Information System (INIS)

Marzali, A.; Schwettman, H.A.

1992-01-01

Microphonics in multi-cell linac structures lead to energy and pointing modulation of the electron beam despite RF stabilization. Evaluation of the microphonic behaviour of a 500 MHz two cell structure is planned in collaboration with Lawrence Berkeley Laboratory and Brookhaven National Laboratory. In this paper we describe a method of evaluation based on accelerometer measurements. (Author) fig., 2 tabs., 5 refs

The effects of asymmetric directional microphone fittings on acceptance of background noise.

Science.gov (United States)

Kim, Jong S; Bryan, Melinda Freyaldenhoven

2011-05-01

The effects of asymmetric directional microphone fittings (i.e., an omnidirectional microphone on one ear and a directional microphone on the other) on speech understanding in noise and acceptance of background noise were investigated in 15 full-time hearing aid users. Subjects were fitted binaurally with four directional microphone conditions (i.e., binaural omnidirectional, right asymmetric directional, left asymmetric directional and binaural directional microphones) using Siemens Intuis Directional behind-the-ear hearing aids. Speech understanding in noise was assessed using the Hearing in Noise Test, and acceptance of background noise was assessed using the Acceptable Noise Level procedure. Speech was presented from 0° while noise was presented from 180° azimuth. The results revealed that speech understanding in noise improved when using asymmetric directional microphones compared to binaural omnidirectional microphone fittings and was not significantly hindered compared to binaural directional microphone fittings. The results also revealed that listeners accepted more background noise when fitted with asymmetric directional microphones as compared to binaural omnidirectional microphones. Lastly, the results revealed that the acceptance of noise was further increased for the binaural directional microphones when compared to the asymmetric directional microphones, maximizing listeners' willingness to accept background noise in the presence of noise. Clinical implications will be discussed.

Statistically optimized near field acoustic holography using an array of pressure-velocity probes

DEFF Research Database (Denmark)

Jacobsen, Finn; Jaud, Virginie

2007-01-01

of a measurement aperture that extends well beyond the source can be relaxed. Both NAH and SONAH are based on the assumption that all sources are on one side of the measurement plane whereas the other side is source free. An extension of the SONAH procedure based on measurement with a double layer array...... of pressure microphones has been suggested. The double layer technique makes it possible to distinguish between sources on the two sides of the array and thus suppress the influence of extraneous noise coming from the “wrong” side. It has also recently been demonstrated that there are significant advantages...... in NAH based on an array of acoustic particle velocity transducers (in a single layer) compared with NAH based on an array of pressure microphones. This investigation combines the two ideas and examines SONAH based on an array of pressure-velocity intensity probes through computer simulations as well...

Irradiation of microphones in the EBR-II core

International Nuclear Information System (INIS)

Gavin, A.P.; Anderson, T.T.; Bobis, J.P.

1976-06-01

Six ANL developed high temperature microphone (acoustic detectors) have been exposed in flowing sodium in the In-Core Instrument Test Facility (INCOT) in the Experimental Breeder Reactor-II (EBR-II) for seven months without any indications of serious degradation of signal output due to the exposure. The YY05 experiment (EBR-II INCOT experiment designation) was performed to obtain data which would be useful in evaluating the ability of the microphones whose active elements are lithium niobate to serve as sensors for acoustic surveillance of fast breeder reactors. The reactor was at full power for 136 days of the experiment exposure period. The microphone temperatures varied from 371 0 C (700 0 F) to 621 0 C (1150 0 F). Neutron exposure varied from 2.64 x 10 22 nvt for the microphone at the elevation of the bottom of the EBR-II core to 0.24 x 10 22 nvt for the microphone at the elevation of the top of an EBR-II fuel assembly. The maximum gamma dose was 5 x 10 12 rads

Near field acoustic holography with microphones mounted on a rigid sphere

DEFF Research Database (Denmark)

Jacobsen, Finn; Moreno, Guillermo; Fernandez Grande, Efren

2008-01-01

Spherical near field acoustic holography (spherical NAH) is a technique that makes it pos-sible to reconstruct the sound field inside and just outside an acoustically transparent spherical surface on which the sound pressure is measured with an array of microphones with negligible scattering...... is only valid if it can be assumed that the sphere has a negligible in-fluence on the incident sound field, and this is not necessarily a good assumption when the sphere is very close to a radiating surface. This paper describes the modified spherical NAH theory and examines the matter through simulations...

General considerations of noise in microphone preamplifiers

NARCIS (Netherlands)

van der Donk, A.G.H.; van der Donk, A.G.H.; Voorthuyzen, J.A.; Voorthuyzen, J.A.; Bergveld, Piet

1991-01-01

In this paper a study of the noise performance of electret microphone systems as a part of hearing aids is presented. The signal-to-noise ratio of the microphone-preamplifier combination, containing a field-effect transistor (FET) and a high value resistive bias element in a hybrid configuration, is

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

International Nuclear Information System (INIS)

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

2010-01-01

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).

On the interference between the two microphones in free-field reciprocity calibration

DEFF Research Database (Denmark)

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

2004-01-01

One of the fundamental assumptions in free-field reciprocity calibration of microphones is that the microphones can be substituted by point sources at the positions where the acoustic centers are located. However, in practice the microphones have finite dimensions and, at the distance and in the ......One of the fundamental assumptions in free-field reciprocity calibration of microphones is that the microphones can be substituted by point sources at the positions where the acoustic centers are located. However, in practice the microphones have finite dimensions and, at the distance...

Micromirror Arrays for Adaptive Optics; TOPICAL

International Nuclear Information System (INIS)

Carr, E.J.

2000-01-01

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

On experimental determination of the random-incidence response of microphones

DEFF Research Database (Denmark)

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

2007-01-01

The random-incidence sensitivity of a microphone is defined as the ratio of the output voltage to the sound pressure that would exist at the position of the acoustic center of the microphone in the absence of the microphone in a sound field with incident plane waves coming from all directions. Th...

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

Directory of Open Access Journals (Sweden)

Yue Kan

2015-06-01

Full Text Available 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.

Factors influencing individual variation in perceptual directional microphone benefit.

Science.gov (United States)

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

2013-01-01

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

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

Science.gov (United States)

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.

Comparison of binaural microphones for externalization of sounds

DEFF Research Database (Denmark)

Cubick, Jens; Sánchez Rodríguez, C.; Song, Wookeun

2015-01-01

or with microphones placed inside the ear canals of a person. In this study, binaural room impulse responses (BRIRs) were measured with several commercially available binaural microphones, both placed inside the listeners’ ears (individual BRIR) and on a head and torso simulator (generic BRIR). The degree...

Real-time adaptive concepts in acoustics blind signal separation and multichannel echo cancellation

CERN Document Server

Schobben, Daniel W E

2001-01-01

Blind Signal Separation (BSS) deals with recovering (filtered versions of) source signals from an observed mixture thereof. The term `blind' relates to the fact that there are no reference signals for the source signals and also that the mixing system is unknown. This book presents a new method for blind signal separation, which is developed to work on microphone signals. Acoustic Echo Cancellation (AEC) is a well-known technique to suppress the echo that a microphone picks up from a loudspeaker in the same room. Such acoustic feedback occurs for example in hands-free telephony and can lead to a perceived loud tone. For an application such as a voice-controlled television, a stereo AEC is required to suppress the contribution of the stereo loudspeaker setup. A generalized AEC is presented that is suited for multi-channel operation. New algorithms for Blind Signal Separation and multi-channel Acoustic Echo Cancellation are presented. A background is given in array signal processing methods, adaptive filter the...

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

Science.gov (United States)

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.

Adaptive Injection-locking Oscillator Array for RF Spectrum Analysis

International Nuclear Information System (INIS)

Leung, Daniel

2011-01-01

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

An investigation of methods for free-field comparison calibration of measurement microphones

DEFF Research Database (Denmark)

Barrera-Figueroa, Salvador; Moreno Pescador, Guillermo; Jacobsen, Finn

2010-01-01

Free-field comparison calibration of measurement microphones requires that a calibrated reference microphone and a test microphone are exposed to the same sound pressure in a free field. The output voltages of the microphones can be measured either sequentially or simultaneously. The sequential...... method requires the sound field to have good temporal stability. The simultaneous method requires instead that the sound pressure is the same in the positions where the microphones are placed. In this paper the results of the application of the two methods are compared. A third combined method...

The static pressure and temperature coefficients of laboratory standard microphones

DEFF Research Database (Denmark)

Rasmussen, Knud

1999-01-01

, for a given type of microphone, can be described by a single function when the coefficients are normalized by their low-frequency value and the frequency is normalized with respect to the individual resonance frequency of the microphone. The theoretical results are supported by experimentally determined...... on an extended lumped parameter representation of the mechanical and acoustic elements of the microphone. The extension involves the frequency dependency of the dynamic diaphragm mass and stiffness as well as a first-order approximation of resonances in the back cavity. It was found that each coefficient...... coefficients for about twenty samples of microphone types B&K 4160 and B&K 4180....

New Technology-Driven Approaches in the Design of Preamplifiers for Condenser Microphones

DEFF Research Database (Denmark)

Haas-Christensen, Jelena

The topic of this thesis is the design of CMOS preamplifiers for condenser microphones. Increasingly popular type of condenser microphones are MEMS (micro-electro-mechanical) microphones which pose a stringent requirements to the design of interface electronics among other due to their increased...... noise. Besides that, as MEMS microphones are easy to integrate with CMOS circuitry, CMOS circuit design gains importance because it can contribute to the overall improved performance of the system by introducing extra functionalities. Possible methods of sensing a signal from the microphone...... of a CMOS interface for a capacitive sensor. Finally, in the fourth part, a novel preamplifier designed demonstrating a concept of differential operation of two microphones biased with voltages of opposite polarities has been described. The amplifier shows how accompanying electronic circuitry can be used...

Fiber Optic Microphone

Science.gov (United States)

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.

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

Directory of Open Access Journals (Sweden)

Rudolf Häusler

2008-06-01

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

Modelling measurement microphones using BEM with visco-thermal losses

DEFF Research Database (Denmark)

Cutanda Henriquez, Vicente; Juhl, Peter Møller

2012-01-01

For many decades, models that can explain the behaviour of measurement condenser microphones have been proposed in the literature. These devices have an apparently simple working principle, a charged capacitor whose charge varies when one of its electrodes, the diaphragm, moves as a result of sound...... waves. However, measurement microphones must be manufactured very carefully due to their sensitivity to small changes of their physical parameters. There are different elements in a microphone, the diaphragm, the gap behind it, a back cavity, a vent for pressure equalization and an external medium. All...... visco-thermal losses is used to model measurement condenser microphones. The models presented are fully coupled and include a FEM model of the diaphragm. The behaviour of the acoustic variables in the gap and the effect of the pressure equalization vent are discussed, as well as the practical difficulty...

Application of optical processing to adaptive phased array radar

Science.gov (United States)

Carroll, C. W.; Vijaya Kumar, B. V. K.

1988-01-01

The results of the investigation of the applicability of optical processing to Adaptive Phased Array Radar (APAR) data processing will be summarized. Subjects that are covered include: (1) new iterative Fourier transform based technique to determine the array antenna weight vector such that the resulting antenna pattern has nulls at desired locations; (2) obtaining the solution of the optimal Wiener weight vector by both iterative and direct methods on two laboratory Optical Linear Algebra Processing (OLAP) systems; and (3) an investigation of the effects of errors present in OLAP systems on the solution vectors.

Dynamic Adaptive Neural Network Arrays: A Neuromorphic Architecture

Energy Technology Data Exchange (ETDEWEB)

Disney, Adam [University of Tennessee (UT); Reynolds, John [University of Tennessee (UT)

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.

Contact microphone using optical fibre Bragg grating technology

International Nuclear Information System (INIS)

Bezombes, F A; Lalor, M J; Burton, D R

2007-01-01

A contact microphone using optical fibre Bragg grating has been developed. It enables one to listen and record a human voice and/or breathing by monitoring the vibration generated by the outer wall of the throat during speech. This system can have many applications such as detecting defects in vocal folds, measuring and monitoring the vibration and defection generated by intubations of a patient throat and other voice related problem, low level speaking recording and transmitting is also possible, the microphone can be also used to monitor breathing and the system can be used as a microphone in very harsh environments for example it would allow one to hear the patient during a cat scan

Efficient voice activity detection in reverberant enclosures using far field microphones

DEFF Research Database (Denmark)

Petsatodis, Theodore; Boukis, Christos

2009-01-01

An algorithm suitable for voice activity detection under reverberant conditions is proposed in this paper. Due to the use of far-filed microphones the proposed solution processes speech signals of highly-varying intensity and signal to noise ratio, that are contaminated with several echoes....... The core of the system is a pair of Hidden Markov Models, that effectively model the speech presence and speech absence situations. To minimise mis-detections an adaptive threshold is used, while a hang-over scheme caters for the intra-frame correlation of speech signals. Experimental results conducted...

Analysis of Acoustic Feedback/Echo Cancellation in Multiple-Microphone and Single-Loudspeaker Systems Using a Power Transfer Function Method

DEFF Research Database (Denmark)

Guo, Meng; Bo Elmedyb, Thomas; Jensen, Søren Holdt

2011-01-01

In this work, we analyze a general multiple-microphone and single-loudspeaker audio processing system, where a multichannel adaptive system is used to cancel the effect of acoustic feedback/echo, and a beamformer processes the feedback/echo canceled signals. We introduce and derive an accurate...

49 CFR 325.73 - Microphone distance correction factors. 1

Science.gov (United States)

2010-10-01

... 49 Transportation 5 2010-10-01 2010-10-01 false Microphone distance correction factors. 1 325.73 Section 325.73 Transportation Other Regulations Relating to Transportation (Continued) FEDERAL MOTOR... MOTOR CARRIER NOISE EMISSION STANDARDS Correction Factors § 325.73 Microphone distance correction...

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

International Nuclear Information System (INIS)

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

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

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

Science.gov (United States)

Chung, King; McKibben, Nicholas

2011-10-01

Wind noise can be a nuisance or a debilitating masker for cochlear implant users in outdoor environments. Previous studies indicated that wind noise at the microphone/hearing aid output had high levels of low-frequency energy and the amount of noise generated is related to the microphone directionality. Currently, cochlear implants only offer either directional microphones or omnidirectional microphones for users at-large. As all cochlear implants utilize pre-emphasis filters to reduce low-frequency energy before the signal is encoded, effective wind noise reduction algorithms for hearing aids might not be applicable for cochlear implants. The purposes of this study were to investigate the effect of microphone directionality on speech recognition and perceived sound quality of cochlear implant users in wind noise and to derive effective wind noise reduction strategies for cochlear implants. A repeated-measure design was used to examine the effects of spectral and temporal masking created by wind noise recorded through directional and omnidirectional microphones and the effects of pre-emphasis filters on cochlear implant performance. A digital hearing aid was programmed to have linear amplification and relatively flat in-situ frequency responses for the directional and omnidirectional modes. The hearing aid output was then recorded from 0 to 360° at flow velocities of 4.5 and 13.5 m/sec in a quiet wind tunnel. Sixteen postlingually deafened adult cochlear implant listeners who reported to be able to communicate on the phone with friends and family without text messages participated in the study. Cochlear implant users listened to speech in wind noise recorded at locations that the directional and omnidirectional microphones yielded the lowest noise levels. Cochlear implant listeners repeated the sentences and rated the sound quality of the testing materials. Spectral and temporal characteristics of flow noise, as well as speech and/or noise characteristics before

Radiation impedance of condenser microphones and their diffuse-field responses

DEFF Research Database (Denmark)

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

2010-01-01

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

Static pressure and temperature coefficients of laboratory standard microphones

DEFF Research Database (Denmark)

Rasmussen, Knud

1996-01-01

of the microphone. The static pressure and temperature coefficients were determined experimentally for about twenty samples of type BK 4160 and BK 4180 microphones. The results agree almost perfectly with the predictions for BK 4160, while some modifications of the lumped parameter values are called for to make......-order approximation of resonances in the back cavity. It was found that each of the coefficients, for a given type of microphone, can be expressed by a single function when the coefficients are normalized by their low-frequency value and the frequency axis normalized by the individual resonance frequency...

Phased array technique for low signal-to-noise ratio wind tunnels, Phase I

Data.gov (United States)

National Aeronautics and Space Administration — Closed wind tunnel beamforming for aeroacoustics has become more and more prevalent in recent years. Still, there are major drawbacks as current microphone arrays...

Development of leak detection system using high temperature-resistant microphones

International Nuclear Information System (INIS)

Morishita, Yoshitsugu; Mochizuki, Hiroyasu; Watanabe, Kenshiu; Nakamura, Takahisa; Nakazima, Yoshiaki; Yamauchi, Tatsuya

1995-01-01

This report describes the development and testing of a coolant leak detection system for an inlet feeder pipe of an advanced thermal reactor (ATR) using high temperature-resistant microphones. Such microphones must be resistant to both high temperatures and high radiation doses. Leakage sound characteristics, attenuation of the sound level in a heat insulating box for the inlet feeder pipes, and background noise were investigated using the experimental facility and the prototype ATR 'FUGEN'. The optimum frequency ranges for the microphone were then determined based on the observed leakage sound and background noise. The ability of the microphone to discriminate between leaks and other burst-type noises was also investigated by statistical analyses. Finally, it was confirmed that the present method could detect a leak within a couple of seconds. (author)

Fabrication of silicon condenser microphones using single wafer technology

NARCIS (Netherlands)

Scheeper, P.R.; van der Donk, A.G.H.; Olthuis, Wouter; Bergveld, Piet

1992-01-01

A condenser microphone design that can be fabricated using the sacrificial layer technique is proposed and tested. The microphone backplate is a 1-¿m plasma-enhanced chemical-vapor-deposited (PECVD) silicon nitride film with a high density of acoustic holes (120-525 holes/mm2), covered with a thin

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

National Research Council Canada - National Science Library

Wage, Kathleen E

2009-01-01

The overall goal of this project is to design adaptive array processing algorithms that have good transient performance, are robust to mismatch, work with low sample support, and incorporate waveguide...

Lumped-parameters equivalent circuit for condenser microphones modeling.

Science.gov (United States)

Esteves, Josué; Rufer, Libor; Ekeom, Didace; Basrour, Skandar

2017-10-01

This work presents a lumped parameters equivalent model of condenser microphone based on analogies between acoustic, mechanical, fluidic, and electrical domains. Parameters of the model were determined mainly through analytical relations and/or finite element method (FEM) simulations. Special attention was paid to the air gap modeling and to the use of proper boundary condition. Corresponding lumped-parameters were obtained as results of FEM simulations. Because of its simplicity, the model allows a fast simulation and is readily usable for microphone design. This work shows the validation of the equivalent circuit on three real cases of capacitive microphones, including both traditional and Micro-Electro-Mechanical Systems structures. In all cases, it has been demonstrated that the sensitivity and other related data obtained from the equivalent circuit are in very good agreement with available measurement data.

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

Science.gov (United States)

Huang, Xun

2009-05-01

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

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

Science.gov (United States)

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.

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

Directory of Open Access Journals (Sweden)

Shang-Yin Lee

2011-01-01

Full Text Available 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.

Preamplifier with ultra low frequency cutoff for infrasonic condenser microphone

DEFF Research Database (Denmark)

Kinnerup, Rasmus Trock; Marbjerg, Kresten; Rasmussen, Per

2012-01-01

low frequencies becomes a challenge. The electric preamplifier presented in this paper together with a prepolarized condenser microphone form a measurement system. The developed preamplifier connects the microphone signal directly to the input of an operational amplifier with ultra high input...

Optical microphone

Energy Technology Data Exchange (ETDEWEB)

Veligdan, J.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.

Design and analysis of diaphragms in dynamic microphones

Directory of Open Access Journals (Sweden)

Zi-Gui Huang

2015-07-01

Full Text Available Most contemporary high-end microphones are dynamic microphones, adopting the most basic electromagnetic transduction principles. This study investigated the diaphragm structures of dynamic microphones. The diaphragms were composed of polyimide material, and the boundary settings required for actual operation were provided using finite element model analysis software. The characteristic frequencies caused by grooving variations on the three-dimensional diaphragm were analyzed for the various groove shapes and number. The groove angles and width variations were examined based on the optimal groove shape selected in the aforementioned analysis, and the effects of these shapes were determined based on the analytical results. Acoustic waves cause thin films to vibrate, forming the working principle behind dynamic microphones. The thin film drives a coil to vibrate in a magnetic field and cuts the line of magnetic force, subsequently producing a voltage on both ends of the coil. This audio-frequency-inducted voltage represents an acoustic wave message. The finite element model analysis software was used to conduct electromagnetic induction simulations; the sound source was fed to the diaphragm to drive the coil. The coil vibrations caused the line of magnetic force to be cut, and the final voltages produced were examined and compared.

Static pressure and temperature coefficients of working standard microphones

DEFF Research Database (Denmark)

Barrera Figueroa, Salvador; Cutanda Henriquez, Vicente; Torras Rosell, Antoni

2016-01-01

be a significant contribution to the uncertainty of the measurement. Determining the environmental coefficients of individual specimens of measurement microphones can be a straightforward though time-consuming procedure provided the appropriate facilities are available. An alternative is to determine them using...... coefficients. For this purpose, the environmental coefficients of some commercially available microphones have been determined experimentally, and whenever possible, compared with the coefficients determined numerically using the Boundary Element Method....... for these coefficients which are used for calibration purposes. Working standard microphones are not exempt of these influences. However, manufacturers usually provide a low frequency value of the environmental coefficient. While in some applications the influence of this coefficient may be negligible, in others it may...

Dynamic Pressure Microphones

Science.gov (United States)

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.

A recurrent neural network for adaptive beamforming and array correction.

Science.gov (United States)

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. Copyright © 2016 Elsevier Ltd. All rights reserved.

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

Science.gov (United States)

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

2009-08-01

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

The fabrication techniques of Z-pinch targets. Techniques of fabricating self-adapted Z-pinch wire-arrays

International Nuclear Information System (INIS)

Qiu Longhui; Wei Yun; Liu Debin; Sun Zuoke; Yuan Yuping

2002-01-01

In order to fabricate wire arrays for use in the Z-pinch physical experiments, the fabrication techniques are investigated as follow: Thickness of about 1-1.5 μm of gold is electroplated on the surface of ultra-fine tungsten wires. Fibers of deuterated-polystyrene (DPS) with diameters from 30 to 100 microns are made from molten DPS. And two kinds of planar wire-arrays and four types of annular wire-arrays are designed, which are able to adapt to the variation of the distance between the cathode and anode inside the target chamber. Furthermore, wire-arrays with diameters form 5-24 μm are fabricated with tungsten wires, respectively. The on-site test shows that the wire-arrays can self-adapt to the distance changes perfectly

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

Science.gov (United States)

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.

Advantages of directional hearing aid microphones related to room acoustics

NARCIS (Netherlands)

Leeuw, A. R.; Dreschler, W. A.

1991-01-01

In this study, two types of hearing aids were used. Both aids had the same frequency characteristics for frontal sound, but one employed an omnidirectional microphone and the other a directional microphone. The frequency characteristics of both hearing aids were measured for five azimuths on KEMAR

Electromagnetic Investigation of a CMOS MEMS Inductive Microphone

Directory of Open Access Journals (Sweden)

Farès TOUNSI

2009-09-01

Full Text Available This paper presents a detailed electromagnetic modeling for a new structure of a monolithic CMOS micromachined inductive microphone. We have shown, that the use of an alternative current (AC in the primary fixed inductor results in a substantially higher induced voltage in the secondary inductor comparing to the case when a direct current (DC is used. The expected increase of the induced voltage can be expressed by a voltage ratio of AC and DC solutions that is in the range of 3 to 6. A prototype fabrication of this microphone has been realized using a combination of standard CMOS 0.6 µm process with a CMOS-compatible post-process consisting in a bulk micromachining technology. The output voltage of the electrodynamic microphone that achieves the µV range can be increased by the use of the symmetric dual-layer spiral inductor structure.

Influence of a remote microphone on localization with hearing aids

DEFF Research Database (Denmark)

Selby, Johan G.; Weisser, Adam; MacDonald, Ewen

2017-01-01

When used with hearing aids (HA), the addition of a remote microphone (RM) may alter the spatial perception of the listener. First, the RM signal is presented diotically from the HAs. Second, the processing in the HA often delays the RM signal relative to the HA microphone signals. Finally...

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

Science.gov (United States)

Dehé, Alfons

2017-06-01

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

A mathematical model for source separation of MMG signals recorded with a coupled microphone-accelerometer sensor pair.

Science.gov (United States)

Silva, Jorge; Chau, Tom

2005-09-01

Recent advances in sensor technology for muscle activity monitoring have resulted in the development of a coupled microphone-accelerometer sensor pair for physiological acousti signal recording. This sensor can be used to eliminate interfering sources in practical settings where the contamination of an acoustic signal by ambient noise confounds detection but cannot be easily removed [e.g., mechanomyography (MMG), swallowing sounds, respiration, and heart sounds]. This paper presents a mathematical model for the coupled microphone-accelerometer vibration sensor pair, specifically applied to muscle activity monitoring (i.e., MMG) and noise discrimination in externally powered prostheses for below-elbow amputees. While the model provides a simple and reliable source separation technique for MMG signals, it can also be easily adapted to other aplications where the recording of low-frequency (< 1 kHz) physiological vibration signals is required.

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

Science.gov (United States)

Bicen, Baris

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

A study into the design of steerable microphone arrays

CERN Document Server

Lai, Chiong Ching; Leung, Yee Hong

2017-01-01

The book covers the design formulations for broadband beamformer targeting nearfield and farfield sources. The book content includes background information on the acoustic environment, including propagation medium, the array geometries, signal models and basic beamformer designs. Subsequently it introduces design formulation for nearfield, farfield and mixed nearfield-farfield beamformers and extends the design formulation into electronically steerable beamformers. In addition, a robust formulation is introduced for all the designs mentioned.

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

Directory of Open Access Journals (Sweden)

Christoph Beck

2016-10-01

Full Text Available 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.

The capture and recreation of 3D auditory scenes

Science.gov (United States)

Li, Zhiyun

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

A Piezoelectric MEMS Microphone Based on Lead Zirconate Titanate (PZT) Thim Films

National Research Council Canada - National Science Library

Polcawich, Ronald

2004-01-01

.... A piezoelectric-based microphone can provide a solution to these requirements, since it offers the ability to passively sense without the power requirements of condenser or piezoresistive microphone counterparts...

Adaptive antenna array algorithms and their impact on code division ...

African Journals Online (AJOL)

In this paper four each blind adaptive array algorithms are developed, and their performance under different test situations (e.g. A WGN (Additive White Gaussian Noise) channel, and multipath environment) is studied A MATLAB test bed is created to show their performance on these two test situations and an optimum one ...

High Dynamic Range adaptive ΔΣ-based Focal Plane Array architecture

KAUST Repository

Yao, Shun; Kavusi, Sam; Salama, Khaled N.

2012-01-01

In this paper, an Adaptive Delta-Sigma based architecture for High Dynamic Range (HDR) Focal Plane Arrays is presented. The noise shaping effect of the Delta-Sigma modulation in the low end, and the distortion noise induced in the high end of Photo

The ribbon microphone: A teaching aid for low frequency electromagnetic education

CSIR Research Space (South Africa)

Van Wyk, Marius S

2017-09-01

Full Text Available The ribbon microphone lends itself as a good example to use for education of multi-physics computer modeling and simulation. The value of the ribbon microphone as teaching aid can be extended by adding a transformer and electronic amplifier...

Investigation of excimer laser ablation threshold of polymers using a microphone

Energy Technology Data Exchange (ETDEWEB)

Krueger, Joerg; Niino, Hiroyuki; Yabe, Akira

2002-09-30

KrF excimer laser ablation of polyethylene terephthalate (PET), polyimide (PI) and polycarbonate (PC) in air was studied by an in situ monitoring technique using a microphone. The microphone signal generated by a short acoustic pulse represented the etch rate of laser ablation depending on the laser fluence, i.e., the ablation 'strength'. From a linear relationship between the microphone output voltage and the laser fluence, the single-pulse ablation thresholds were found to be 30 mJ cm{sup -2} for PET, 37 mJ cm{sup -2} for PI and 51 mJ cm{sup -2} for PC (20-pulses threshold). The ablation thresholds of PET and PI were not influenced by the number of pulses per spot, while PC showed an incubation phenomenon. A microphone technique provides a simple method to determine the excimer laser ablation threshold of polymer films.

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

Science.gov (United States)

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

2018-03-01

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

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

Science.gov (United States)

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.

On Acoustic Feedback Cancellation Using Probe Noise in Multiple-Microphone and Single-Loudspeaker Systems

DEFF Research Database (Denmark)

Guo, Meng; Elmedyb, Thomas Bo; Jensen, Søren Holdt

2012-01-01

of the adaptive estimation is significantly decreased when keeping the steady-state error unchanged. The goal of this work is to derive analytic expressions for the system behavior such as convergence rate and steady-state error for a multiple-microphone and single-loudspeaker audio system, where the acoustic...... feedback cancellation is carried out using a probe noise signal. The derived results show how different system parameters and signal properties affect the cancellation performance, and the results explain theoretically the decreased convergence rate. Understanding this is important for making further...

Simulation Study of Electronic Damping of Microphonic Vibrations in Superconducting Cavities

International Nuclear Information System (INIS)

Alicia Hofler; Jean Delayen

2005-01-01

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

The optimal configuration of photovoltaic module arrays based on adaptive switching controls

International Nuclear Information System (INIS)

Chao, Kuei-Hsiang; Lai, Pei-Lun; Liao, Bo-Jyun

2015-01-01

Highlights: • We propose a strategy for determining the optimal configuration of a PV array. • The proposed strategy was based on particle swarm optimization (PSO) method. • It can identify the optimal module array connection scheme in the event of shading. • It can also find the optimal connection of a PV array even in module malfunctions. - Abstract: This study proposes a strategy for determining the optimal configuration of photovoltaic (PV) module arrays in shading or malfunction conditions. This strategy was based on particle swarm optimization (PSO). If shading or malfunctions of the photovoltaic module array occur, the module array immediately undergoes adaptive reconfiguration to increase the power output of the PV power generation system. First, the maximal power generated at various irradiation levels and temperatures was recorded during normal array operation. Subsequently, the irradiation level and module temperature, regardless of operating conditions, were used to recall the maximal power previously recorded. This previous maximum was compared with the maximal power value obtained using the maximum power point tracker to assess whether the PV module array was experiencing shading or malfunctions. After determining that the array was experiencing shading or malfunctions, PSO was used to identify the optimal module array connection scheme in abnormal conditions, and connection switches were used to implement optimal array reconfiguration. Finally, experiments were conducted to assess the strategy for identifying the optimal reconfiguration of a PV module array in the event of shading or malfunctions

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

Science.gov (United States)

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.

Towards an enhanced performance of uniform circular arrays at low frequencies

DEFF Research Database (Denmark)

Tiana Roig, Elisabet; Torras Rosell, Antoni; Fernandez Grande, Efren

2013-01-01

are mounted on a scatterer such as a rigid cylinder or a sphere. The beamforming output improves with increasing frequency, up to a certain frequency where spatial aliasing occurs. At low frequencies the performance is limited by the radius of the array; in other words, given a certain number of microphones...

Environmental coefficients of the free-field sensitivity of measurement microphones

DEFF Research Database (Denmark)

Barrera Figueroa, Salvador; Cutanda Henriquez, Vicente; Torras-Rosell, Antoni

2017-01-01

The sensitivity of measurement microphones, both pressure and free field, is affected by changes in the environmental conditions, mainly temperature and static pressure. Static pressure and temperature coefficients for the pressure sensitivity have been the object of previous studies focused...... on Laboratory Standard microphones and few working standard microphones. The literature describes frequency dependent values for these coefficients which are used for calibration purposes. However, there is no description of the environmental coefficients of the free-field sensitivity though there have been...... some implementations that attempt to take care of the differences between the coefficients for the two types of sensitivities. Measuring the coefficients in a free field poses some challeng; it is not so easy to change neither the static pressure nor the temperature inside anechoic room within...

Acoustic isolation vessel for measurement of the background noise in microphones

Science.gov (United States)

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.

A time-selective technique for free-field reciprocity calibration of condenser microphones

DEFF Research Database (Denmark)

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

2003-01-01

In normal practice, microphones are calibrated in a closed coupler where the sound pressure is uniformly distributed over the diaphragm. Alternatively, microphones can be placed in a free field, although in that case the distribution of sound pressure over the diaphragm will change as a result of...

Characteristics of Relocated Quiet Zones Using Virtual Microphone Algorithm in an Active Headrest System

Directory of Open Access Journals (Sweden)

Seokhoon Ryu

2016-01-01

Full Text Available This study displays theoretical and experimental investigation on the characteristics of the relocated zone of quiet by a virtual microphone (VM based filtered-x LMS (FxLMS algorithm which can be embedded in a real-time digital controller for an active headrest system. The attenuation changes at the relocated zones of quiet by the variation of the distance between the ear and the error microphone are mainly examined. An active headrest system was implemented for the control experiment at a chair and consists of two (left and right secondary loudspeakers, two error microphones, two observer microphones at ear positions in a HATS, and other electronics including a dSPACE 1401 controller. The VM based FxLMS algorithm achieved an attenuation of about 22 dB in the control experiment against a narrowband primary noise by the variation of the distance between the ear and the error microphone. The important factors for the algorithm are discussed as well.

On determination of microphone response and other parameters by a hybrid experimental and numerical method

DEFF Research Database (Denmark)

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

2008-01-01

to this problem is to measure the velocity distribution of the membrane by means of a non-contact method, such as laser vibrometry. The measured velocity distributions can be used together with a numerical formulation such as the Boundary Element Method for estimating the microphone response and other parameters...... such as the acoustic centres. In this work, a hybrid method is presented. The velocity distributions of condenser Laboratory Standard microphones were measured using a laser vibrometer. This measured velocity distribution was used for estimating the microphone responses and parameters. The agreement with experimental......Typically, numerical calculations of the pressure, free-field and random-incidence response of a condenser microphone are carried out on the basis of an assumed displacement distribution of the diaphragm of the microphone; the conventional assumption is that the displacement follows a Bessel...

Feasible pickup from intact ossicular chain with floating piezoelectric microphone.

Science.gov (United States)

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

2012-02-22

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

Feasible pickup from intact ossicular chain with floating piezoelectric microphone

Directory of Open Access Journals (Sweden)

Kang Hou-Yong

2012-02-01

Full Text Available Abstract 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.

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

Science.gov (United States)

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

2017-11-01

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

MEMS capacitive accelerometer-based middle ear microphone.

Science.gov (United States)

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

2012-12-01

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

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

Science.gov (United States)

Moatti, L; Busquet, D; Cotin, G

1983-01-01

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

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

Science.gov (United States)

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

77 FR 64446 - Wireless Microphones Proceeding

Science.gov (United States)

2012-10-22

... to balance the needs of potential new classes of wireless microphone licensees with those of other... undercut that balance by significantly reducing the amount of spectrum available for other uses, such as by..., a space capacity-rated for 3,000 people; for sports venues, a minimum of 10,000 seats for indoors...

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

Directory of Open Access Journals (Sweden)

Hailong Xu

2016-03-01

Full Text Available 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.

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

DEFF Research Database (Denmark)

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

2009-01-01

to this problem is to measure the velocity distribution of the membrane by means of a non-contact method, such as laser vibrometry. The measured velocity distribution can be used together with a numerical formulation such as the boundary element method for estimating the microphone response and other parameters......, e.g., the acoustic center. In this work, such a hybrid method is presented and examined. The velocity distributions of a number of condenser microphones have been determined using a laser vibrometer, and these measured velocity distributions have been used for estimating microphone responses......Typically, numerical calculations of the pressure, free-field, and random-incidence response of a condenser microphone are carried out on the basis of an assumed displacement distribution of the diaphragm of the microphone; the conventional assumption is that the displacement follows a Bessel...

Optical wave microphone measurement during laser ablation of Si

Energy Technology Data Exchange (ETDEWEB)

Mitsugi, Fumiaki, E-mail: mitsugi@cs.kumamoto-u.ac.jp [Graduate School of Science and Technology, Kumamoto University, 2-39-1 Kurokami, Kumamoto, 860-8555 (Japan); Ide, Ryota; Ikegami, Tomoaki [Graduate School of Science and Technology, Kumamoto University, 2-39-1 Kurokami, Kumamoto, 860-8555 (Japan); Nakamiya, Toshiyuki; Sonoda, Yoshito [Graduate School of Industrial Engineering, Tokai University, 9-1-1 Toroku, Kumamoto, 862-8652 (Japan)

2012-10-30

Pulsed laser irradiation is used for surface treatment of a solid and ablation for particle formation in gas, liquid or supercritical phase media. When a pulsed laser is used to irradiate a solid, spatial refractive index variations (including photothermal expansion, shockwaves and particles) occur, which vary depending on the energy density of the pulsed laser. We focused on this phenomenon and applied an unique method for detection of refractive index variation using an optical wave microphone based on Fraunhofer diffraction. In this research, we analyzed the waveforms and frequencies of refractive index variations caused by pulsed laser irradiation of silicon in air and measured with an optical wave microphone.

DFT-Domain Based Single-Microphone Noise Reduction for Speech Enhancement

DEFF Research Database (Denmark)

C. Hendriks, Richard; Gerkmann, Timo; Jensen, Jesper

As speech processing devices like mobile phones, voice controlled devices, and hearing aids have increased in popularity, people expect them to work anywhere and at any time without user intervention. However, the presence of acoustical disturbances limits the use of these applications, degrades...... their performance, or causes the user difficulties in understanding the conversation or appreciating the device. A common way to reduce the effects of such disturbances is through the use of single-microphone noise reduction algorithms for speech enhancement. The field of single-microphone noise reduction...

Optical wave microphone measurements of laser ablation of copper in supercritical carbon dioxide

Energy Technology Data Exchange (ETDEWEB)

Mitsugi, Fumiaki, E-mail: mitsugi@cs.kumamoto-u.ac.jp [Graduate School of Science and Technology, Kumamoto University, 2-39-1 Kurokami, Kumamoto 860-8555 (Japan); Ikegami, Tomoaki [Graduate School of Science and Technology, Kumamoto University, 2-39-1 Kurokami, Kumamoto 860-8555 (Japan); Nakamiya, Toshiyuki; Sonoda, Yoshito [Graduate School of Industrial Engineering, Tokai University, 9-1-1 Toroku, Kumamoto 862-8652 (Japan)

2013-11-29

Laser ablation plasma in a supercritical fluid has attracted much attention recently due to its usefulness in forming nanoparticles. Observation of the dynamic behavior of the supercritical fluid after laser irradiation of a solid is necessary for real-time monitoring and control of laser ablation. In this study, we utilized an optical wave microphone to monitor pulsed laser irradiation of a solid in a supercritical fluid. The optical wave microphone works based on Fraunhofer diffraction of phase modulation of light by changes in refractive index. We hereby report on our measurements for pulsed laser irradiation of a Cu target in supercritical carbon dioxide using an optical wave microphone. Photothermal acoustic waves which generated after single pulsed laser irradiation of a Cu target were detectable in supercritical carbon dioxide. The speed of sound around the critical point of supercritical carbon dioxide was clearly slower than that in gas. The optical wave microphone detected a signal during laser ablation of Cu in supercritical carbon dioxide that was caused by shockwave degeneration. - Highlights: • Photothermal acoustic wave in supercritical fluid was observed. • Sound speed around the critical point was slower than that in gas. • Optical wave microphone detected degeneration of a shockwave. • Ablation threshold of a solid in supercritical fluid can be estimated. • Generation of the second shockwave in supercritical phase was suggested.

Using adaptive antenna array in LTE with MIMO for space-time processing

Directory of Open Access Journals (Sweden)

Abdourahamane Ahmed Ali

2015-04-01

Full Text Available The actual methods of improvement the existent wireless transmission systems are proposed. Mathematical apparatus is considered and proved by models, graph of which are shown, using the adaptive array antenna in LTE with MIMO for space-time processing. The results show that improvements, which are joined with space-time processing, positively reflects on LTE cell size or on throughput

Stand-Off Chemical Detection Using Photoacoustic Sensing Techniques—From Single Element to Phase Array

Directory of Open Access Journals (Sweden)

Deepa Gupta

2018-01-01

Full Text Available Technologies that can detect harmful chemicals, such as explosive devices, harmful gas leaks, airborne chemicals or/and biological agents, are heavily invested in by the government to prevent any possible catastrophic consequences. Some key features of such technology are, but not limited to, effective signal-to-noise ratio (SNR of the detected signal and extended distance between the detector and target. In this work, we describe the development of photoacoustic sensing techniques from simple to more complex systems. These techniques include passive and active noise filters, parabolic sound reflectors, a lock-in amplifier, and beam-forming with an array of microphones; using these techniques, we increased detection distance from a few cm in an indoor setting to over 41 feet in an outdoor setting. We also establish a theoretical mathematical model that explains the underlying principle of how SNR can be improved with an increasing number of microphone elements in the phase array. We validate this model with computational simulations as well as experimental results.

An Accurate Study on Capacitive Microphone with Circular Diaphragm Using a Higher Order Elasticity Theory

Directory of Open Access Journals (Sweden)

Shakiba Dowlati

Full Text Available Abstract This study has been undertaken to investigate the mechanical behavior of the capacitive microphone with clamped circular diaphragm using modified couple stress theory in comparison to the classical one. Presence of the length scale parameter in modified couple stress theory provides the means to evaluate the size effect on the microphone mechanical behavior. Investigating Pull-in phenomenon and dynamic behavior of the microphone are the matters provided due to the application of a step DC voltage. Also the effects of different air damping coefficients on dynamic pull-in voltage and pull-in time have been studied. The output level or sensitivity of the microphone has been studied by investigating the frequency response in term of magnitude for different length scale parameters to figure out how the length scale parameter affects on the sensitivity of the capacitive microphone. To achieve these ends, the nonlinear differential equation of the circular diaphragm has been extracted using Kirchhoff thin plate theory. Then, a Step-by-Step Linearization Method (SSLM has been used to escape from the nonlinearity of the differential equation. Afterwards, Galerkin-based reduced-order model has been applied to solve the obtained equation.

High Dynamic Range adaptive ΔΣ-based Focal Plane Array architecture

KAUST Repository

Yao, Shun

2012-10-16

In this paper, an Adaptive Delta-Sigma based architecture for High Dynamic Range (HDR) Focal Plane Arrays is presented. The noise shaping effect of the Delta-Sigma modulation in the low end, and the distortion noise induced in the high end of Photo-diode current were analyzed in detail. The proposed architecture can extend the DR for about 20N log2 dB at the high end of Photo-diode current with an N bit Up-Down counter. At the low end, it can compensate for the larger readout noise by employing Extended Counting. The Adaptive Delta-Sigma architecture employing a 4-bit Up-Down counter achieved about 160dB in the DR, with a Peak SNR (PSNR) of 80dB at the high end. Compared to the other HDR architectures, the Adaptive Delta-Sigma based architecture provides the widest DR with the best SNR performance in the extended range.

Radar techniques using array antennas

CERN Document Server

Wirth, Wulf-Dieter

2013-01-01

Radar Techniques Using Array Antennas is a thorough introduction to the possibilities of radar technology based on electronic steerable and active array antennas. Topics covered include array signal processing, array calibration, adaptive digital beamforming, adaptive monopulse, superresolution, pulse compression, sequential detection, target detection with long pulse series, space-time adaptive processing (STAP), moving target detection using synthetic aperture radar (SAR), target imaging, energy management and system parameter relations. The discussed methods are confirmed by simulation stud

Adaptive port-starboard beamforming of triplet arrays

NARCIS (Netherlands)

Beerens, S.P.; Been, R.; Groen, J.; Noutary, E.; Doisy, Y.

2000-01-01

Triplet arrays are single line arrays with three hydrophones on a circular section of the array. The triplet structure provides immediate port-starboard (PS) discrimination. This paper discusses the theoretical and experimental performance of triplet arrays. Results are obtained on detection gain

Improved multi-microphone noise reduction preserving binaural cues

NARCIS (Netherlands)

Koutrouvelis, A.; Hendriks, R.C.; Jensen, J; Heusdens, R.; Dong, Min; Zheng, Thomas Fang

2016-01-01

We propose a new multi-microphone noise reduction technique for binaural cue preservation of the desired source and the interferers. This method is based on the linearly constrained minimum variance (LCMV) framework, where the constraints are used for the binaural cue preservation of the desired

Shooter Localization in Wireless Microphone Networks

Directory of Open Access Journals (Sweden)

David Lindgren

2010-01-01

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

Parametric Investigation of Laser Doppler Microphones

Science.gov (United States)

Daoud, M.; Naguib, A.

2002-11-01

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

Numerical design and testing of a sound source for secondary calibration of microphones using the Boundary Element Method

DEFF Research Database (Denmark)

Cutanda Henriquez, Vicente; Juhl, Peter Møller; Barrera Figueroa, Salvador

2009-01-01

Secondary calibration of microphones in free field is performed by placing the microphone under calibration in an anechoic chamber with a sound source, and exposing it to a controlled sound field. A calibrated microphone is also measured as a reference. While the two measurements are usually made...... apart to avoid acoustic interaction. As a part of the project Euromet-792, aiming to investigate and improve methods for secondary free-field calibration of microphones, a sound source suitable for simultaneous secondary free-field calibration has been designed using the Boundary Element Method...... of the Danish Fundamental Metrology Institute (DFM). The design and verification of the source are presented in this communication....

Reproducibility of Dual-Microphone Voice Range Profile Equipment

DEFF Research Database (Denmark)

Printz, Trine; Pedersen, Ellen Raben; Juhl, Peter

2017-01-01

in an anechoic chamber and an office: (a) comparing sound pressure levels (SPLs) from a dual-microphone VRP device, the Voice Profiler, when given the same input repeatedly (test-retest reliability); (b) comparing SPLs from 3 devices when given the same input repeatedly (intervariation); and (c) assessing...

Proceedings of the Adaptive Sensor Array Processing Workshop (12th) Held in Lexington, MA on 16-18 March 2004 (CD-ROM)

National Research Council Canada - National Science Library

James, F

2004-01-01

...: The twelfth annual workshop on Adaptive Sensor Array Processing presented a diverse agenda featuring new work on adaptive methods for communications, radar and sonar, algorithmic challenges posed...

Filtering microphonics in dark matter germanium experiments

International Nuclear Information System (INIS)

Morales, J.; Garcia, E.; Ortiz de Solorzano, A.; Morales, A.; Nunz-Lagos, R.; Puimedon, J.; Saenz, C.; Villar, J.A.

1992-01-01

A technique for reducing the microphonic noise in a germanium spectrometer used in dark matter particles searches is described. Filtered energy spectra, corresponding to 48.5 kg day of data in a running experiment in the Canfranc tunnel are presented. Improvements of this filtering procedure with respect to the method of rejecting those events not distributed evenly in time are also discussed. (orig.)

Multichannel signal enhancement using a remote wireless microphone

NARCIS (Netherlands)

Bloemendal, Brian; Van De Laar, Jakob; Sommen, Piet

2012-01-01

A novel approach to multichannel signal enhancement is presented that exploits data from a remote wireless microphone (RWM). This RWM is placed near an interfering source and transmits only autocorrelation data of its observations to a host, i.e., not the entire signal. The host has access to the

Video Conferencing for a Virtual Seminar Room

DEFF Research Database (Denmark)

Forchhammer, Søren; Fosgerau, A.; Hansen, Peter Søren K.

2002-01-01

A PC-based video conferencing system for a virtual seminar room is presented. The platform is enhanced with DSPs for audio and video coding and processing. A microphone array is used to facilitate audio based speaker tracking, which is used for adaptive beam-forming and automatic camera...

Development and assessment of two fixed-array microphones for use with hearing aids

NARCIS (Netherlands)

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

1993-01-01

Hearing-impaired listeners often have great difficulty understanding speech in situations with background noise (e.g., meetings, parties) . Conventional hearing aids offer insufficient directivity to significantly reduce background noise relative to the desired speech signal . Based on array

The Microphone Feedback Analogy for Chatter in Machining

Directory of Open Access Journals (Sweden)

Tony Schmitz

2015-01-01

Full Text Available This paper provides experimental evidence for the analogy between the time-delay feedback in public address systems and chatter in machining. Machining stability theory derived using the Nyquist criterion is applied to predict the squeal frequency in a microphone/speaker setup. Comparisons between predictions and measurements are presented.

Wavefront sensing and adaptive control in phased array of fiber collimators

Science.gov (United States)

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

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

Science.gov (United States)

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

2017-03-01

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

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

Directory of Open Access Journals (Sweden)

Paul Kendrick

Full Text Available 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.

Chip-size-packaged silicon microphones [for hearing instruments

DEFF Research Database (Denmark)

Müllenborn, Matthias; Rombach, Pirmin; Klein, Udo

2001-01-01

bonding. The devices are fully encapsulated and provided with a well-determined interface to the environment. The integrated microphones operate at a bias of 1.5 V and are expected to reach a sensitivity of 5 mV/Pa, an A-weighted equivalent input noise of 24 dB sound pressure level, and a power...

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

Science.gov (United States)

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.

Regularised reconstruction of sound fields with a spherical microphone array

DEFF Research Database (Denmark)

Granados Corsellas, Alba; Jacobsen, Finn; Fernandez Grande, Efren

2013-01-01

implementation might lead to disastrous reconstructions. A large number of regularisation tools based on singular value decomposition are available, and it has been found that the acoustic holography problem for certain geometries can be formulated in such a way that similarities to singular value decomposition...... become apparent. Hence, a number of regularisation methods, including truncated singular value decomposition, standard Tikhonov, constrained Tikhonov, iterative Tikhonov, Landweber and Rutishauser, have been adapted for spherical near field acoustic holography. The accuracy of the methods is examined...

Development of early core anomaly detection system by using in-sodium microphone in JOYO. Fundamental characteristics test of in-sodium microphone in water and examination of improvement of detection accuracy

International Nuclear Information System (INIS)

Komai, Masafumi

2001-07-01

Fast reactor core anomalies can be detected in near real-time with acoustic sensors. An acoustic detection system senses an in-core anomaly immediately from the fast acoustic signals that propagate through the sodium coolant. One example of a detectable anomaly is sodium boiling due to local blockage in a sub-assembly; the slight change in background acoustic signals can be detected. A key advantage of the acoustic detector is that it can be located outside the core. The location of the anomaly in the core can be determined by correlating multiple acoustic signals. This report describes the testing and fundamental characteristics of a microphone suitable for use in the sodium coolant and examines methods to improve the system's S/N ratio. Testing in water confirmed that the in-sodium microphone has good impulse and wide band frequency responses. These tests used impulse and white noise signals that imitate acoustic signals from boiling sodium. Correlation processing of multiple microphone signals to improve S/N ratio is also described. (author)

Blind source extraction for a combined fixed and wireless sensor network

NARCIS (Netherlands)

Bloemendal, B.B.A.J.; Laar, van de J.; Sommen, P.C.W.

2012-01-01

The emergence of wireless microphones in everyday life creates opportunities to exploit spatial diversity when using fixed microphone arrays combined with these wireless microphones. Traditional array signal processing (ASP) techniques are not suitable for such a scenario since the locations of the

Microphone detected ionacoustic signal from metals

International Nuclear Information System (INIS)

Dioszeghy, T.; Szoekefalvi-Nagy, Z.; Biro, T.

1986-12-01

An experimental system for studying the radiation-induced acoustic signal generated by a modulated 2 MeV He + ion beam in metals is described. For detection, a closed cell on the rear side of the copper or aluminium sample, a half-inch condenser microphone, and a lock-in amplifier were employed. The signal was found to be proportional to beam current and particle energy, and inversely proportional to cell length. A decrease of the signal magnitude and an increase of the phase delay with increasing modulation frequency and sample thickness were also observed. (author)

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

Energy Technology Data Exchange (ETDEWEB)

Guianvarc' h, Cecile; Pitre, Laurent [Laboratoire Commun de Metrologie LNE/Cnam, 61 rue du Landy, 93210 La Plaine Saint Denis (France); Gavioso, Roberto M.; Benedetto, Giuliana [Istituto Nazionale di Ricerca Metrologica, Strada delle Cacce 91, 10135 Turin (Italy); Bruneau, Michel [Laboratoire d' Acoustique de l' Universite du Maine UMR CNRS 6613, av. Olivier Messiaen, 72085 Le Mans Cedex 9 (France)

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.

Improved prediction error filters for adaptive feedback cancellation in hearing aids

DEFF Research Database (Denmark)

Ngo, Kim; van Waterschoot, Toon; Christensen, Mads Græsbøll

2013-01-01

feedback cancellation (AFC) where the goal is to adaptively model the acoustic feedback path and estimate the feedback signal, which is then subtracted from the microphone signal. The main problem in identifying the acoustic feedback path model is the correlation between the near-end signal...

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

Directory of Open Access Journals (Sweden)

Chia-Chang Hu

2005-04-01

Full Text Available 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 (J, the rank of the MWF (M, the system processing gain (N, and the number of samples in a chip interval (S, that is, 𝒪(JMNS. The complexity of the equivalent detector based on the minimum mean-squared error (MMSE or the subspace-based eigenstructure analysis is a function of 𝒪((JNS3. 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 J-element antenna array, the amount of the L-sample support, and the rank of the M-stage MWF. The performance advantage of the proposed detector over other DS-CDMA detectors is investigated as well.

A Comparison of Acoustic Field Measurement by a Microphone and by an Optical Interferometric Probe

Directory of Open Access Journals (Sweden)

R. Bálek

2002-01-01

Full Text Available The objective of this work is to show that our optical method for measuring acoustic pressure is in some way superior to measurement using a microphone. Measurement of the integral acoustic pressure in the air by a laser interferometric probe is compared with measurement using a microphone. We determined the particular harmonic components in the acoustic field in the case of relatively high acoustic power in the ultrasonic frequency range.

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

Science.gov (United States)

Tan, Chee Wee; Miao, Jianmin

2009-06-01

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

Prediction of Quadcopter State through Multi-Microphone Side-Channel Fusion

NARCIS (Netherlands)

Koops, Hendrik Vincent; Garg, Kashish; Kim, Munsung; Li, Jonathan; Volk, Anja; Franchetti, Franz

Improving trust in the state of Cyber-Physical Systems becomes increasingly important as more tasks become autonomous. We present a multi-microphone machine learning fusion approach to accurately predict complex states of a quadcopter drone in flight from the sound it makes using audio content

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

Science.gov (United States)

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.

A low-noise/low-power preamplifier for capacitive microphones

DEFF Research Database (Denmark)

Fürst, Claus Erdmann

1996-01-01

A design for a microphone preamplifier for application in hearing aids is presented. The amplifier operates at a supply of 1-1.5 V, the current drain is 40 μA. The maximum sound level allowed is more than 120 dB SPL (Sound Pressure Level), with a typical noise level of 25 dB(A) SPL (A...

MP.EXE Microphone pressure sensitivity calibration calculation program

DEFF Research Database (Denmark)

Rasmussen, Knud

1999-01-01

MP.EXE is a program which calculates the pressure sensitivity of LS1 microphones as defined in IEC 61094-1, based on measurement results performed as laid down in IEC 61094-2.A very early program was developed and written by K. Rasmussen. The code of the present heavily extended version is writte...... by E.S. Olsen.The present manual is written by K.Rasmussen and E.S. Olsen....

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

Science.gov (United States)

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

2016-01-01

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

Virtual microphone sensing through vibro-acoustic modelling and Kalman filtering

Science.gov (United States)

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

2018-05-01

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

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

Science.gov (United States)

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

1996-06-01

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

Far-Field Voice Activity Detection and Its Applications in Adverse Acoustic Environments

DEFF Research Database (Denmark)

Petsatodis, Theodoros

2012-01-01

-sided Gamma distribution. The increased adaptability of the system along with the encapsulated adaptive threshold allows the system to perform remarkably under adverse complex phenomena. Following recent technological trends, of incorporating microphone arrays in numerous commercial applications (eg. mobile...... phones, VOIP terminals) and research environments (smart rooms), a multiple microphone VAD is also considered. The system processes signals captured by far-field sensors in order to integrate spatial information in addition to the frequency content available at a single sensor. The core of the system......-modality of speech production, a simple visual-VAD is also developed to examine performance enhancement when fusing audio and video information. In the final part of the work, applications of VAD in the context of integration with other signal processing systems are also considered. Performance benefits of combining...

Adaptive motion compensation in sonar array processing

NARCIS (Netherlands)

Groen, J.

2006-01-01

In recent years, sonar performance has mainly improved via a significant increase in array ap-erture, signal bandwidth and computational power. This thesis aims at improving sonar array processing techniques based on these three steps forward. In applications such as anti-submarine warfare and mine

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

Science.gov (United States)

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.

Method for discriminating microphonic noise in proportional counters

International Nuclear Information System (INIS)

Gold, R.

1991-01-01

This patent describes a detector system responsive to nuclear events for installation in a downhole logging tool including measuring well drilling equipment which subjects the detection system to microphonic shock. It comprises a closed chamber subject to impinging nuclear events and having two separate anode wires therein spaced apart from each other and spanning the chamber, providing a pair of separated spaced output terminals to thereby form an output signal; circuit means connecting from at least one of the chamber output terminals to a different amplifier means having two input terminals; the circuit means connected from the output terminal of the chamber to one of the input terminals of the differential amplifier means to cause formation of an output signal from the differential amplifier means; and vibration shock responsive means mounted in the detector system and having an output terminal which forms an output signal for connection to a second input at the differential circuit means so that microphonic signals from the chamber and the shock responsive means are provided thereto and tend to cancel when applied to the input terminals thereof, and wherein the shock responsive means does not cancel at the differential circuit means signals relating to nuclear events from the detector system

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

National Research Council Canada - National Science Library

Bulbuller, Gokhan

2006-01-01

.... In this study, a speech recognition system is presented, specifically an isolated word recognizer which uses speech collected from the external auditory canals of the subjects via an in-ear microphone...

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

Energy Technology Data Exchange (ETDEWEB)

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

Development of a leak detection system using high temperature-resistant microphones

International Nuclear Information System (INIS)

Morishita, Yoshitsugu; Mochizuki, Hiroyasu; Watanabe, Kenshiu; Nakamura, Takahisa; Nakajima, Yoshiaki; Yamauchi, Tatsuya

1991-01-01

This report describes the development of a detection system of coolant leak from an inlet feeder pipe of an Advanced Thermal Reactor (ATR) with high temperature-resistant microphones. A microphone having resistance to both high temperature and high radiation dose has been developed at first. The characteristics with regard to leakage sound, attenuation of sound level in a heat insulating box for the inlet feeder pipes and background noise were clarified by laboratory experiments and measurements in the prototype ATR 'Fugen'. On the basis of these experimental findings, appropriate frequency ranges were surveyed to detect the leakage sound with a high S/N ratio under the background noise. Reliability of the system to a malfunction caused by burst-type noises observed in the plant was also investigated by statistical analyses. Finally, it was confirmed that the present method could detect a leak within a couple of seconds. (author)

Performance Analysis of Blind Beamforming Algorithms in Adaptive Antenna Array in Rayleigh Fading Channel Model

International Nuclear Information System (INIS)

Yasin, M; Akhtar, Pervez; Pathan, Amir Hassan

2013-01-01

In this paper, we analyze the performance of adaptive blind algorithms – i.e. Kaiser Constant Modulus Algorithm (KCMA), Hamming CMA (HAMCMA) – with CMA in a wireless cellular communication system using digital modulation technique. These blind algorithms are used in digital signal processor of adaptive antenna to make it smart and change weights of the antenna array system dynamically. The simulation results revealed that KCMA and HAMCMA provide minimum mean square error (MSE) with 1.247 dB and 1.077 dB antenna gain enhancement, 75% reduction in bit error rate (BER) respectively over that of CMA. Therefore, KCMA and HAMCMA algorithms give a cost effective solution for a communication system

Design and performance evaluation of a broadband three dimensional acoustic intensity measuring system.

Science.gov (United States)

Miah, Khalid H; Hixon, Elmer L

2010-04-01

A seven-microphone three dimensional (3D) intensity measuring system has been developed and evaluated for performance for a broad frequency band (200 Hz-6.5 kHz). Six microphones are arranged in a concentric array with one microphone at the center of the probe. The screw adjustable center microphone is the probe reference microphone, and is used for calibrations of the other microphones in the probe. This probe addresses limitations of the traditional two-microphone system in measuring acoustical properties in a 3D space from the one dimensional measurements. This probe also eliminates the need of spacers used in the existing 3D probes for broadband measurements. Diffraction and reflection effects on calibrations due to presence of the microphones and the probe supporting structure are negligible. This seven-microphone probe provided better results in the intensity measurements for the wide frequency band than that of a similar four-microphone array probe.

MP.EXE, a Calculation Program for Pressure Reciprocity Calibration of Microphones

DEFF Research Database (Denmark)

Rasmussen, Knud

1998-01-01

A computer program is described which calculates the pressure sensitivity of microphones based on measurements of the electrical transfer impedance in a reciprocity calibration set-up. The calculations are performed according to the International Standard IEC 6194-2. In addition a number of options...

Remote Microphone System Use at Home: Impact on Caregiver Talk

Science.gov (United States)

Benítez-Barrera, Carlos R.; Angley, Gina P.; Tharpe, Anne Marie

2018-01-01

Purpose: The purpose of this study was to investigate the effects of home use of a remote microphone system (RMS) on the spoken language production of caregivers with young children who have hearing loss. Method: Language Environment Analysis recorders were used with 10 families during 2 consecutive weekends (RMS weekend and No-RMS weekend). The…

Mitigating Wind Induced Noise in Outdoor Microphone Signals Using a Singular Spectral Subspace Method

Directory of Open Access Journals (Sweden)

Omar Eldwaik

2018-01-01

Full Text Available Wind induced noise is one of the major concerns of outdoor acoustic signal acquisition. It affects many field measurement and audio recording scenarios. Filtering such noise is known to be difficult due to its broadband and time varying nature. In this paper, a new method to mitigate wind induced noise in microphone signals is developed. Instead of applying filtering techniques, wind induced noise is statistically separated from wanted signals in a singular spectral subspace. The paper is presented in the context of handling microphone signals acquired outdoor for acoustic sensing and environmental noise monitoring or soundscapes sampling. The method includes two complementary stages, namely decomposition and reconstruction. The first stage decomposes mixed signals in eigen-subspaces, selects and groups the principal components according to their contributions to wind noise and wanted signals in the singular spectrum domain. The second stage reconstructs the signals in the time domain, resulting in the separation of wind noise and wanted signals. Results show that microphone wind noise is separable in the singular spectrum domain evidenced by the weighted correlation. The new method might be generalized to other outdoor sound acquisition applications.

Visualization of Broadband Sound Sources

OpenAIRE

Sukhanov Dmitry; Erzakova Nadezhda

2016-01-01

In this paper the method of imaging of wideband audio sources based on the 2D microphone array measurements of the sound field at the same time in all the microphones is proposed. Designed microphone array consists of 160 microphones allowing to digitize signals with a frequency of 7200 Hz. Measured signals are processed using the special algorithm that makes it possible to obtain a flat image of wideband sound sources. It is shown experimentally that the visualization is not dependent on the...

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

Science.gov (United States)

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

2006-01-01

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

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

Science.gov (United States)

Fisher, Aileen

The term infrasound describes atmospheric sound waves with frequencies below 20 Hz, while acoustics are classified within the audible range of 20 Hz to 20 kHz. Infrasound and acoustic monitoring in the scientific community is hampered by low signal-to-noise ratios and a limited number of studies on regional and short-range noise and source characterization. The JASON Report (2005) suggests the infrasound community focus on more broad-frequency, observational studies within a tactical distance of 10 km. In keeping with that recommendation, this paper presents a study of regional and short-range atmospheric acoustic and infrasonic noise characterization, at a desert site in West Texas, covering a broad frequency range of 0.2 to 100 Hz. To spatially sample the band, a large number of infrasound gauges was needed. A laboratory instrument analysis is presented of the set of low-cost infrasound sensors used in this study, manufactured by Inter-Mountain Laboratories (IML). Analysis includes spectra, transfer functions and coherences to assess the stability and range of the gauges, and complements additional instrument testing by Sandia National Laboratories. The IMLs documented here have been found reliably coherent from 0.1 to 7 Hz without instrument correction. Corrections were built using corresponding time series from the commercially available and more expensive Chaparral infrasound gauge, so that the corrected IML outputs were able to closely mimic the Chaparral output. Arrays of gauges are needed for atmospheric sound signal processing. Our West Texas experiment consisted of a 1.5 km aperture, 23-gauge infrasound/acoustic array of IMLs, with a compact, 12 m diameter grid-array of rented IMLs at the center. To optimize signal recording, signal-to-noise ratio needs to be quantified with respect to both frequency band and coherence length. The higher-frequency grid array consisted of 25 microphones arranged in a five by five pattern with 3 meter spacing, without

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

Science.gov (United States)

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.

Mechanical performance of SiC based MEMS capacitive microphone for ultrasonic detection in harsh environment

Science.gov (United States)

Zawawi, S. A.; Hamzah, A. A.; Mohd-Yasin, F.; Majlis, B. Y.

2017-08-01

In this project, SiC based MEMS capacitive microphone was developed for detecting leaked gas in extremely harsh environment such as coal mines and petroleum processing plants via ultrasonic detection. The MEMS capacitive microphone consists of two parallel plates; top plate (movable diaphragm) and bottom (fixed) plate, which separated by an air gap. While, the vent holes were fabricated on the back plate to release trapped air and reduce damping. In order to withstand high temperature and pressure, a 1.0 μm thick SiC diaphragm was utilized as the top membrane. The developed SiC could withstand a temperature up to 1400°C. Moreover, the 3 μm air gap is invented between the top membrane and the bottom plate via wafer bonding. COMSOL Multiphysics simulation software was used for design optimization. Various diaphragms with sizes of 600 μm2, 700 μm2, 800 μm2, 900 μm2 and 1000 μm2 are loaded with external pressure. From this analysis, it was observed that SiC microphone with diaphragm width of 1000 μm2 produced optimal surface vibrations, with first-mode resonant frequency of approximately 36 kHz. The maximum deflection value at resonant frequency is less than the air gap thickness of 8 mu;m, thus eliminating the possibility of shortage between plates during operation. As summary, the designed SiC capacitive microphone has high potential and it is suitable to be applied in ultrasonic gas leaking detection in harsh environment.

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

Science.gov (United States)

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

Millimeter-Wave Microstrip Antenna Array Design and an Adaptive Algorithm for Future 5G Wireless Communication Systems

Directory of Open Access Journals (Sweden)

Cheng-Nan Hu

2016-01-01

Full Text Available This paper presents a high gain millimeter-wave (mmW low-temperature cofired ceramic (LTCC microstrip antenna array with a compact, simple, and low-profile structure. Incorporating minimum mean square error (MMSE adaptive algorithms with the proposed 64-element microstrip antenna array, the numerical investigation reveals substantial improvements in interference reduction. A prototype is presented with a simple design for mass production. As an experiment, HFSS was used to simulate an antenna with a width of 1 mm and a length of 1.23 mm, resonating at 38 GHz. Two identical mmW LTCC microstrip antenna arrays were built for measurement, and the center element was excited. The results demonstrated a return loss better than 15 dB and a peak gain higher than 6.5 dBi at frequencies of interest, which verified the feasibility of the design concept.

Distributed 3D Source Localization from 2D DOA Measurements Using Multiple Linear Arrays

Directory of Open Access Journals (Sweden)

Antonio Canclini

2017-01-01

Full Text Available This manuscript addresses the problem of 3D source localization from direction of arrivals (DOAs in wireless acoustic sensor networks. In this context, multiple sensors measure the DOA of the source, and a central node combines the measurements to yield the source location estimate. Traditional approaches require 3D DOA measurements; that is, each sensor estimates the azimuth and elevation of the source by means of a microphone array, typically in a planar or spherical configuration. The proposed methodology aims at reducing the hardware and computational costs by combining measurements related to 2D DOAs estimated from linear arrays arbitrarily displaced in the 3D space. Each sensor measures the DOA in the plane containing the array and the source. Measurements are then translated into an equivalent planar geometry, in which a set of coplanar equivalent arrays observe the source preserving the original DOAs. This formulation is exploited to define a cost function, whose minimization leads to the source location estimation. An extensive simulation campaign validates the proposed approach and compares its accuracy with state-of-the-art methodologies.

Leak detection in the primary reactor coolant piping of nuclear power plant by applying beam-microphone technology

International Nuclear Information System (INIS)

Kasai, Yoshimitsu; Shimanskiy, Sergey; Naoi, Yosuke; Kanazawa, Junichi

2004-01-01

A microphone leak detection method was applied to the inlet piping of the ATR-prototype reactor, Fugen. Statistical analysis results showed that the cross-correlation method provided the effective results for detection of a small leakage. However, such a technique has limited application due to significant distortion of the signals on the reactor site. As one of the alternative methods, the beam-microphone provides necessary spatial selectivity and its performance is less affected by signal distortion. A prototype of the beam-microphone was developed and then tested at the O-arai Engineering Center of the Japan Nuclear Cycle Development Institute (JNC). On-site testing of the beam-microphone was carried out in the inlet piping room of an RBMK reactor of the Leningrad Nuclear Power Plant (LNPP) in Russia. A leak sound imitator was used to simulate the leakage sound under the leakage flow condition of 1-3 gpm (0.23-0.7 m 3 /h). Analysis showed that signal distortion does not seriously affect the performance of this method, and that sound reflection may result in the appearance of ghost sound sources. The test results showed that the influences of sound reflection and background noise were smaller at the high frequencies where the leakage location could be estimated with an angular accuracy of 5deg which is the range of localization accuracy required for the leak detection system. (author)

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

Science.gov (United States)

Kyriakou, Kyriaki; Fisher, Helene R.

2013-01-01

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

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

Science.gov (United States)

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

2018-04-01

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

Noise-Canceling Helmet Audio System

Science.gov (United States)

Seibert, Marc A.; Culotta, Anthony J.

2007-01-01

A prototype helmet audio system has been developed to improve voice communication for the wearer in a noisy environment. The system was originally intended to be used in a space suit, wherein noise generated by airflow of the spacesuit life-support system can make it difficult for remote listeners to understand the astronaut s speech and can interfere with the astronaut s attempt to issue vocal commands to a voice-controlled robot. The system could be adapted to terrestrial use in helmets of protective suits that are typically worn in noisy settings: examples include biohazard, fire, rescue, and diving suits. The system (see figure) includes an array of microphones and small loudspeakers mounted at fixed positions in a helmet, amplifiers and signal-routing circuitry, and a commercial digital signal processor (DSP). Notwithstanding the fixed positions of the microphones and loudspeakers, the system can accommodate itself to any normal motion of the wearer s head within the helmet. The system operates in conjunction with a radio transceiver. An audio signal arriving via the transceiver intended to be heard by the wearer is adjusted in volume and otherwise conditioned and sent to the loudspeakers. The wearer s speech is collected by the microphones, the outputs of which are logically combined (phased) so as to form a microphone- array directional sensitivity pattern that discriminates in favor of sounds coming from vicinity of the wearer s mouth and against sounds coming from elsewhere. In the DSP, digitized samples of the microphone outputs are processed to filter out airflow noise and to eliminate feedback from the loudspeakers to the microphones. The resulting conditioned version of the wearer s speech signal is sent to the transceiver.

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

Science.gov (United States)

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

2018-03-23

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

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

Science.gov (United States)

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

2011-01-01

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

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

Science.gov (United States)

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

2016-01-01

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

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

Science.gov (United States)

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

2017-05-01

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

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

Science.gov (United States)

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

2018-05-01

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

Application of phased array technology for identification of low frequency noise sources

Energy Technology Data Exchange (ETDEWEB)

Hugo E. Camargo; Patricio A. Ravetta; Ricardo A. Burdisso; Adam K. Smith [NIOSH (United States)

2009-12-15

A study conducted by the National Institute for Occupational Safety and Health (NIOSH) revealed that 90% of coal miners have hearing impairment by age 50, compared to only 10% of those not exposed to occupational noise. According to the Mine Safety and Health Administration (MSHA), Continuous Mining Machine (CM) operators account for 30% of workers exposed to noise doses exceeding the Permissible Exposure Level (PEL). In this context, NIOSH is conducting research to identify and control dominant noise sources in CMs. Previous noise source identification was performed using a Bruel & Kjaer (B&K) 1.92-m diameter, 42-microphone phased array. These measurements revealed that the impacts from the conveyor chain onto the tail roller, and the impacts from the conveyor chain onto the upper deck are the dominant noise sources at the tail-section of the CM. The objectives of the work presented in this paper were: (1) To rank the noise radiated by the different sections of the conveyor, and (2) to determine the effect of a urethane-coated tail roller on the noise radiated by the tail-section. This test was conducted using an Acoustical and Vibrations Engineering Consultants (AVEC) 3.5-m diameter, 121-microphone phased array. The results from this new test show that a urethane-coated tail roller yields reductions in the tail-section of 2 to 8 dB in Sound Pressure Level in the frequency range of 1 kHz to 5 kHz. However, integration of the acoustic maps shows that the front-section and mid-section of the conveyor also contain dominant noise sources. Therefore, a urethane-coated tail roller in combination with a chain with urethane-coated flights that reduces the noise sources in the front and mid sections of the conveyor is required to yield a significant noise reduction on the CM operator's overall exposure. These results show the applicability of phased array technology for low frequency noise source identification.

Bit-rate reduction strategies for noise suppression with a remote wireless microphone

NARCIS (Netherlands)

Cvijanovic, N.; Sadiq, O.; Srinivasan, S.

2012-01-01

In single channel non-stationary noise reduction it is paramount that a good noise reference is available in a timely manner to maintaina high quality speech signal. Using a remote wireless microphone placed close to a noise source, a good estimate of the noise power spectral density (PSD) can be

Bit rate reduction strategies for noise suppression using a remote wireless microphone

NARCIS (Netherlands)

Cvijanovic, N.; Sadiq, O.; Srinivasan, S.

2012-01-01

In single-channel non-stationary noise reduction it is paramount that a good noise reference is available in a timely manner to maintain a high quality speech signal. Using a remote wireless microphone placed close to a noise source, a good estimate of the noise power spectral density (PSD) can be

Compensating microphonics in SRF cavities to ensure beam stability for future free electron lasers

Energy Technology Data Exchange (ETDEWEB)

Neumann, Axel

2008-07-21

In seeded High-Gain-Harmonic-Generation free electron lasers or energy recovery linear accelerators the requirements for the bunch-to-bunch timing and energy jitter of the beam are in the femtosecond and per mill regime. This implies the ability to control the cavity radiofrequency (RF) field to an accuracy of 0.02 in phase and up to 1.10{sup -4} in amplitude. For the planned BESSY-FEL it is envisaged to operate 144 superconducting 1.3 GHz cavities of the 2.3 GeV driver linac in continuous wave mode and at a low beam current. The cavity resonance comprises a very narrow bandwidth of the order of tens of Hertz. Such cavities have been characterized under accelerator like conditions in the HoBiCaT test facility. It was possible to measure the error sources affecting the field stability in continuous wave (CW) operation. Microphonics, the main error source for a mechanical detuning of the cavities, lead to an average fluctuation of the cavity resonance of 1-5 Hz rms. Furthermore, the static and dynamic Lorentz force detuning and the helium pressure dependance of the cavity resonance have been measured. Single cavity RF control and linac bunch-to-bunch longitudinal phase space modeling containing the measured properties showed, that it is advisable to find means to minimize the microphonics detuning by mechanical tuning. Thus, several fast tuning systems have been tested for CW operation. These tuners consist of a motor driven lever for slow and coarse tuning and a piezo that is integrated into the tuner support for fast and fine tuning. Regarding the analysis of the detuning spectrum an adaptive feedforward method based on the least-mean-square filter algorithm has been developed for fast cavity tuning. A detuning compensation between a factor of two and up to a factor of seven has been achieved. Modeling the complete system including the fast tuning scheme, showed that the requirements of the BESSY-FEL are attainable. (orig.)

Compensating microphonics in SRF cavities to ensure beam stability for future free electron lasers

International Nuclear Information System (INIS)

Neumann, Axel

2008-01-01

In seeded High-Gain-Harmonic-Generation free electron lasers or energy recovery linear accelerators the requirements for the bunch-to-bunch timing and energy jitter of the beam are in the femtosecond and per mill regime. This implies the ability to control the cavity radiofrequency (RF) field to an accuracy of 0.02 in phase and up to 1.10 -4 in amplitude. For the planned BESSY-FEL it is envisaged to operate 144 superconducting 1.3 GHz cavities of the 2.3 GeV driver linac in continuous wave mode and at a low beam current. The cavity resonance comprises a very narrow bandwidth of the order of tens of Hertz. Such cavities have been characterized under accelerator like conditions in the HoBiCaT test facility. It was possible to measure the error sources affecting the field stability in continuous wave (CW) operation. Microphonics, the main error source for a mechanical detuning of the cavities, lead to an average fluctuation of the cavity resonance of 1-5 Hz rms. Furthermore, the static and dynamic Lorentz force detuning and the helium pressure dependance of the cavity resonance have been measured. Single cavity RF control and linac bunch-to-bunch longitudinal phase space modeling containing the measured properties showed, that it is advisable to find means to minimize the microphonics detuning by mechanical tuning. Thus, several fast tuning systems have been tested for CW operation. These tuners consist of a motor driven lever for slow and coarse tuning and a piezo that is integrated into the tuner support for fast and fine tuning. Regarding the analysis of the detuning spectrum an adaptive feedforward method based on the least-mean-square filter algorithm has been developed for fast cavity tuning. A detuning compensation between a factor of two and up to a factor of seven has been achieved. Modeling the complete system including the fast tuning scheme, showed that the requirements of the BESSY-FEL are attainable. (orig.)

High frequency microphone measurements for transition detection on airfoils. NACA-0015 appendix report

DEFF Research Database (Denmark)

Døssing, Mads

Time series of pressure fluctuations has been obtained using high frequency microphones distributed over the surface of airfoils undergoing wind tunnel tests in the LM Windtunnel, owned by ’LM Glasfiber’, Denmark. The present report describes the dataanalysis, with special attention given to tran...

Light Dependent Resistance as a Sensor in Spectroscopy Setups Using Pulsed Light and Compared with Electret Microphones

Directory of Open Access Journals (Sweden)

Daniel Acosta-Avalos

2006-05-01

Full Text Available Light-dependent resistances (LDR are cheap light sensors. A less known lightdetector is the electret microphone, whose electret membrane functions as a perfectabsorber, but only detects pulsed light. The aim of this study was to analyze the use of aLDR and an electret microphone as a light sensor in an optical spectroscopy system usingpulsed light. A photoacoustic spectroscopy setup was used, substituting the photoacousticchamber by the light sensor proposed. The absorption spectra of two different liquids wereanalyzed. The results obtained allow the recommendation of the LDR as the first choice inthe construction of cheap homemade pulsed light spectroscopy systems.

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

Science.gov (United States)

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

2018-05-01

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

Measurements of noise immission from wind turbines at receptor locations: Use of a vertical microphone board to improve the signal-to-noise ratio

International Nuclear Information System (INIS)

Fegeant, Olivier

1999-01-01

The growing interest in wind energy has increased the need of accuracy in wind turbine noise immission measurements and thus, the need of new measurement techniques. This paper shows that mounting the microphone on a vertical board improves the signal-to-noise ratio over the whole frequency range compared to the free microphone technique. Indeed, the wind turbine is perceived two times noisier by the microphone due to the signal reflection by the board while, in addition, the wind noise is reduced. Furthermore, the board shielding effect allows the measurements to be carried out in the presence of reflecting surfaces such as building facades

A Multifunction Low-Power Preamplifier for MEMS Capacitive Microphones

DEFF Research Database (Denmark)

Jawed, Syed Arsalan; Nielsen, Jannik Hammel; Gottardi, Massimo

2009-01-01

A multi-function two-stage chopper-stabilized preamplifier (PAMP) for MEMS capacitive microphones (MCM) is presented. The PAMP integrates digitally controllable gain, high-pass filtering and offset control, adding flexibility to the front-end readout of MCMs. The first stage of the PAMP consists...... of a source-follower (SF) while the second-stage is a capacitive gain stage. The second-stage employs chopper-stabilization (CHS), while SF buffer shields the MCM sensor from the switching spurs. The PAMP uses M poly bias resistors for the second-stage, exploiting Miller effect to achieve flat audio...

Visualization of Broadband Sound Sources

Directory of Open Access Journals (Sweden)

Sukhanov Dmitry

2016-01-01

Full Text Available In this paper the method of imaging of wideband audio sources based on the 2D microphone array measurements of the sound field at the same time in all the microphones is proposed. Designed microphone array consists of 160 microphones allowing to digitize signals with a frequency of 7200 Hz. Measured signals are processed using the special algorithm that makes it possible to obtain a flat image of wideband sound sources. It is shown experimentally that the visualization is not dependent on the waveform, but determined by the bandwidth. Developed system allows to visualize sources with a resolution of up to 10 cm.

Microphonics detuning compensation in 3.9 GHZ superconducting RF cavities

International Nuclear Information System (INIS)

Ruben Carcagno

2003-01-01

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

Microphonics detuning compensation in 3.9 GHZ superconducting RF cavities

Energy Technology Data Exchange (ETDEWEB)

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.

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

Science.gov (United States)

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

2013-01-01

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

Active cancellation of probing in linear dipole phased array

CERN Document Server

Singh, Hema; Jha, Rakesh Mohan

2015-01-01

In this book, a modified improved LMS algorithm is employed for weight adaptation of dipole array for the generation of beam pattern in multiple signal environments. In phased arrays, the generation of adapted pattern according to the signal scenario requires an efficient adaptive algorithm. The antenna array is expected to maintain sufficient gain towards each of the desired source while at the same time suppress the probing sources. This cancels the signal transmission towards each of the hostile probing sources leading to active cancellation. In the book, the performance of dipole phased array is demonstrated in terms of fast convergence, output noise power and output signal-to-interference-and noise ratio. The mutual coupling effect and role of edge elements are taken into account. It is established that dipole array along with an efficient algorithm is able to maintain multilobe beamforming with accurate and deep nulls towards each probing source. This work has application to the active radar cross secti...

Adaptive lesion formation using dual mode ultrasound array system

Science.gov (United States)

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

Microphone triggering circuit for elimination of mechanically induced frequency-jitter in diode laser spectrometers: implications for quantitative analysis.

Science.gov (United States)

Sams, R L; Fried, A

1987-09-01

An electronic timing circuit using a microphone triggering device has been developed for elimination of mechanically induced frequency-jitter in diode laser spectrometers employing closed-cycle refrigerators. Mechanical compressor piston shocks are detected by the microphone and actuate an electronic circuit which ultimately interrupts data acquisition until the mechanical vibrations are completely quenched. In this way, laser sweeps contaminated by compressor frequency-jitter are not co-averaged. Employing this circuit, measured linewidths were in better agreement with that calculated. The importance of eliminating this mechanically induced frequency-jitter when carrying out quantitative diode laser measurements is further discussed.

Probe suppression in conformal phased array

CERN Document Server

Singh, Hema; Neethu, P S

2017-01-01

This book considers a cylindrical phased array with microstrip patch antenna elements and half-wavelength dipole antenna elements. The effect of platform and mutual coupling effect is included in the analysis. The non-planar geometry is tackled by using Euler's transformation towards the calculation of array manifold. Results are presented for both conducting and dielectric cylinder. The optimal weights obtained are used to generate adapted pattern according to a given signal scenario. It is shown that array along with adaptive algorithm is able to cater to an arbitrary signal environment even when the platform effect and mutual coupling is taken into account. This book provides a step-by-step approach for analyzing the probe suppression in non-planar geometry. Its detailed illustrations and analysis will be a useful text for graduate and research students, scientists and engineers working in the area of phased arrays, low-observables and stealth technology.

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

Science.gov (United States)

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

2017-12-01

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

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

Science.gov (United States)

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

2017-05-01

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

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

Directory of Open Access Journals (Sweden)

Tharoeun Thap

2016-08-01

Full Text Available 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.

Spiral-Shaped Piezoelectric MEMS Cantilever Array for Fully Implantable Hearing Systems

Directory of Open Access Journals (Sweden)

Péter Udvardi

2017-10-01

Full Text Available Fully implantable, self-powered hearing aids with no external unit could significantly increase the life quality of patients suffering severe hearing loss. This highly demanding concept, however, requires a strongly miniaturized device which is fully implantable in the middle/inner ear and includes the following components: frequency selective microphone or accelerometer, energy harvesting device, speech processor, and cochlear multielectrode. Here we demonstrate a low volume, piezoelectric micro-electromechanical system (MEMS cantilever array which is sensitive, even in the lower part of the voice frequency range (300–700 Hz. The test array consisting of 16 cantilevers has been fabricated by standard bulk micromachining using a Si-on-Insulator (SOI wafer and aluminum nitride (AlN as a complementary metal-oxide-semiconductor (CMOS and biocompatible piezoelectric material. The low frequency and low device footprint are ensured by Archimedean spiral geometry and Si seismic mass. Experimentally detected resonance frequencies were validated by an analytical model. The generated open circuit voltage (3–10 mV is sufficient for the direct analog conversion of the signals for cochlear multielectrode implants.

Free-field reciprocity calibration of laboratory standard (LS) microphones using a time selective technique

DEFF Research Database (Denmark)

Rasmussen, Knud; Barrera Figueroa, Salvador

2006-01-01

Although the basic principle of reciprocity calibration of microphones in a free field is simple, the practical problems are complicated due to the low signal-to-noise ratio and the influence of cross talk and reflections from the surroundings. The influence of uncorrelated noise can be reduced...

Virtual Microphones for Multichannel Audio Resynthesis

Directory of Open Access Journals (Sweden)

Athanasios Mouchtaris

2003-09-01

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

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

Science.gov (United States)

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…

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

International Nuclear Information System (INIS)

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)

Radiation-induced adaptive response in fetal mice: a micro-array study

International Nuclear Information System (INIS)

Vares, G.; Bing, Wang; Mitsuru, Nenoi; Tetsuo, Nakajima; Kaoru, Tanaka; Isamu, Hayata

2006-01-01

Exposure of sublethal doses of ionizing radiation can induce protective mechanisms against a subsequent higher dose irradiation. This phenomenon called radio-adaptation (or adaptive response - AR), has been described in a wide range of biological models. In a series of studies, we demonstrated the existence of a radiation-induced AR in mice during late organogenesis. For better understanding of molecular mechanisms underlying AR in our model, we performed a global analysis of transcriptome regulations in cells collected from whole mouse fetuses. Using cDNA micro-arrays, we studied gene expression in these cells after in utero priming exposure to irradiation. Several combinations of radiation dose and dose-rate were applied to induce or not an AR in our system. Gene regulation was observed after exposure to priming radiation in each condition. Student's t-test was performed in order to identify genes whose expression modulation was specifically different in AR-inducing an( non-AR-inducing conditions. Genes were ranked according to their ability in discriminating AR-specific modulations. Since AR genes were implicated in variety of functions and cellular processes, we applied a functional classification algorithm, which clustered genes in a limited number of functionally related group: We established that AR genes are significantly enriched for specific keywords. Our results show a significant modulation of genes implicated in signal transduction pathways. No AR-specific alteration of DNA repair could be observed. Nevertheless, it is likely that modulation of DNA repair activity results, at least partly, from post-transcriptional regulation. One major hypothesis is that de-regulations of signal transduction pathways and apoptosis may be responsible for AR phenotype. In previous work, we demonstrated that radiation-induced AR in mice during organogenesis is related to Trp53 gene status and to the occurrence of radiation-induced apoptosis. Other work proposed that p53

A low-voltage silicon condenser microphone for hearing instrument applications

DEFF Research Database (Denmark)

Rombach, Pirmin; Müllenborn, Matthias; Klein, Udo

1999-01-01

the input-related noise of the following preamplifier stage becomes dominant and results in a high equivalent input-related noise. Here a silicon condenser microphone with the potential for hearing instrument applications will be presented. To get the best properties for the different mechanical parts, e...... related A-weighted noise is 23 dB SPL, including the preamplifier. Due to a conservative layout, the parasitic capacitance is about 50%. An increase of 2–3 mV/Pa sensitivity and hence 3 dB SPL less noise can therefore be achieved by design optimization....

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

Science.gov (United States)

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

2014-07-01

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

Temperature compensated, humidity insensitive, high-T_g TOPAS FBGs for accelerometers and microphones

DEFF Research Database (Denmark)

Stefani, Alessio; Yuan, W.; Markos, C.

2012-01-01

In this paper we present our latest work on Fiber Bragg Gratings (FBGs) in microstructured polymer optical fibers (mPOFs) and their application as strain sensing transducers in devices, such as accelerometers and microphones. We demonstrate how the cross-sensitivity of the FBG to temperature...

A note on determination of the diffuse-field sensitivity of microphones using the reciprocity technique

DEFF Research Database (Denmark)

Barrera Figueroa, Salvador; Jacobsen, Finn

2008-01-01

angles of incidence but also on the accuracy of the frequency response at normal incidence. By contrast, this paper is concerned with determining the absolute diffuse-field response of a microphone using the reciprocity technique. To examine this possibility, a reciprocity calibration setup is used...

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

Science.gov (United States)

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

2017-09-01

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

Impact of Antenna Placement on Frequency Domain Adaptive Antenna Array in Hybrid FRF Cellular System

Directory of Open Access Journals (Sweden)

Sri Maldia Hari Asti

2012-01-01

Full Text Available Frequency domain adaptive antenna array (FDAAA is an effective method to suppress interference caused by frequency selective fading and multiple-access interference (MAI in single-carrier (SC transmission. However, the performance of FDAAA receiver will be affected by the antenna placement parameters such as antenna separation and spread of angle of arrival (AOA. On the other hand, hybrid frequency reuse can be adopted in cellular system to improve the cellular capacity. However, optimal frequency reuse factor (FRF depends on the channel propagation and transceiver scheme as well. In this paper, we analyze the impact of antenna separation and AOA spread on FDAAA receiver and optimize the cellular capacity by using hybrid FRF.

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

Science.gov (United States)

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

2018-04-01

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

Adaptive Beamforming Based on Complex Quaternion Processes

Directory of Open Access Journals (Sweden)

Jian-wu Tao

2014-01-01

Full Text Available Motivated by the benefits of array signal processing in quaternion domain, we investigate the problem of adaptive beamforming based on complex quaternion processes in this paper. First, a complex quaternion least-mean squares (CQLMS algorithm is proposed and its performance is analyzed. The CQLMS algorithm is suitable for adaptive beamforming of vector-sensor array. The weight vector update of CQLMS algorithm is derived based on the complex gradient, leading to lower computational complexity. Because the complex quaternion can exhibit the orthogonal structure of an electromagnetic vector-sensor in a natural way, a complex quaternion model in time domain is provided for a 3-component vector-sensor array. And the normalized adaptive beamformer using CQLMS is presented. Finally, simulation results are given to validate the performance of the proposed adaptive beamformer.

Limitations of Phased Array Beamforming in Open Rotor Noise Source Imaging

Science.gov (United States)

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

2013-01-01

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

Adaptive radar resource management

CERN Document Server

Moo, Peter

2015-01-01

Radar Resource Management (RRM) is vital for optimizing the performance of modern phased array radars, which are the primary sensor for aircraft, ships, and land platforms. Adaptive Radar Resource Management gives an introduction to radar resource management (RRM), presenting a clear overview of different approaches and techniques, making it very suitable for radar practitioners and researchers in industry and universities. Coverage includes: RRM's role in optimizing the performance of modern phased array radars The advantages of adaptivity in implementing RRMThe role that modelling and

Environmental photobioreactor array (EPBRA) systems and apparatus related thereto

Science.gov (United States)

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

2017-11-14

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

Adaptive smart simulator for characterization and MPPT construction of PV array

International Nuclear Information System (INIS)

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

2016-01-01

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

Adaptive smart simulator for characterization and MPPT construction of PV array

Science.gov (United States)

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

2016-07-01

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

Adaptive smart simulator for characterization and MPPT construction of PV array

Energy Technology Data Exchange (ETDEWEB)

Ouada, Mehdi, E-mail: mehdi.ouada@univ-annaba.org; Meridjet, Mohamed Salah [Electromechanical engineering department, Electromechanical engineering laboratory, Badji Mokhtar University, B.P. 12, Annaba (Algeria); Dib, Djalel [Department of Electrical Engineering, University of Tebessa, Tebessa (Algeria)

2016-07-25

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

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

International Nuclear Information System (INIS)

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

2007-01-01

This paper presents the results of cw RF frequency control and RF phase-stabilization experiments performed with a piezoelectric fast tuner mechanically coupled to a superconducting, 345 MHz, < = 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 = 0.5 triple-spoke-loaded cavity RF field amplitude and phase errors were controlled to ±0.5% and ±30 respectively.

High frequency microphone measurements for transition detection on airfoils. Risø C2-18 appendix report

DEFF Research Database (Denmark)

Døssing, Mads

Time series of pressure fluctuations has been obtained using high frequency microphones distributed over the surface of airfoils undergoing wind tunnel tests in the LM Windtunnel, owned by ’LM Glasfiber’, Denmark. The present report describes the dataanalysis, with special attention given to tran...

High frequency microphone measurements for transition detection on airfoils. Risø B1-18 appendix report

DEFF Research Database (Denmark)

Døssing, Mads

Time series of pressure fluctuations has been obtained using high frequency microphones distributed over the surface of airfoils undergoing wind tunnel tests in the LM Windtunnel, owned by ’LM Glasfiber’, Denmark. The present report describes the dataanalysis, with special attention given to tran...

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

Science.gov (United States)

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

2012-07-01

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

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

Directory of Open Access Journals (Sweden)

Cheng Wang

2016-12-01

Full Text Available 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.

Cold plasma decontamination using flexible jet arrays

Science.gov (United States)

Konesky, Gregory

2010-04-01

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

Adaptive Port-Starboard Beamforming of Triplet Sonar Arrays

NARCIS (Netherlands)

Groen, J.; Beerens, S.P.; Been, R.; Doisy, Y.

2005-01-01

Abstract—For a low-frequency active sonar (LFAS) with a triplet receiver array, it is not clear in advance which signal processing techniques optimize its performance. Here, several advanced beamformers are analyzed theoretically, and the results are compared to experimental data obtained in sea

Time-domain beamforming and blind source separation speech input in the car environment

CERN Document Server

Bourgeois, Julien

2009-01-01

The development of computer and telecommunication technologies led to a revolutioninthewaythatpeopleworkandcommunicatewitheachother.One of the results is that large amount of information will increasingly be held in a form that is natural for users, as speech in natural language. In the presented work, we investigate the speech signal capture problem, which includes the separation of multiple interfering speakers using microphone arrays. Adaptive beamforming is a classical approach which has been developed since the seventies. However it requires a double-talk detector (DTD) that interrupts th

A Simple Approach in Estimating the Effectiveness of Adapting Mirror Concentrator and Tracking Mechanism for PV Arrays in the Tropics

Directory of Open Access Journals (Sweden)

M. E. Ya’acob

2014-01-01

Full Text Available Mirror concentrating element and tracking mechanism has been seriously investigated and widely adapted in solar PV technology. In this study, a practical in-field method is conducted in Serdang, Selangor, Malaysia, for the two technologies in comparison to the common fixed flat PV arrays. The data sampling process is measured under stochastic weather characteristics with the main target of calculating the effectiveness of PV power output. The data are monitored, recorded, and analysed in real time via GPRS online monitoring system for 10 consecutive months. The analysis is based on a simple comparison of the actual daily power generation from each PV generator with statistical analysis of multiple linear regression (MLR and analysis of variance test (ANOVA. From the analysis, it is shown that tracking mechanism generates approximately 88 Watts (9.4% compared to the mirror concentrator which generates 144 Watts (23.4% of the cumulative dc power for different array configurations at standard testing condition (STC references. The significant increase in power generation shows feasibilities of implying both mechanisms for PV generators and thus contributes to additional reference in PV array design.

Localisation d'une source sonore par un réseau de microphones

OpenAIRE

Thaljaoui , Adel; Brulin , Damien; Val , Thierry; Nasri , Nejah

2014-01-01

National audience; L'assistance à domicile d'une personne âgée, notamment la connaissance de sa position géographique en tout instant, est devenue actuellement l'une des problématiques les plus urgentes. L'exploitation de l'information audio captée par un réseau de capteurs munis de microphones constitue un axe de recherche prometteur qui pourrait contribuer à une meilleure localisation dans le cadre des maisons intelligentes. Nous introduisons, dans cet article, nos premiers travaux sur la l...

Field-Deployable Acoustic Digital Systems for Noise Measurement

Science.gov (United States)

Shams, Qamar A.; Wright, Kenneth D.; Lunsford, Charles B.; Smith, Charlie D.

2000-01-01

Langley Research Center (LaRC) has for years been a leader in field acoustic array measurement technique. Two field-deployable digital measurement systems have been developed to support acoustic research programs at LaRC. For several years, LaRC has used the Digital Acoustic Measurement System (DAMS) for measuring the acoustic noise levels from rotorcraft and tiltrotor aircraft. Recently, a second system called Remote Acquisition and Storage System (RASS) was developed and deployed for the first time in the field along with DAMS system for the Community Noise Flight Test using the NASA LaRC-757 aircraft during April, 2000. The test was performed at Airborne Airport in Wilmington, OH to validate predicted noise reduction benefits from alternative operational procedures. The test matrix was composed of various combinations of altitude, cutback power, and aircraft weight. The DAMS digitizes the acoustic inputs at the microphone site and can be located up to 2000 feet from the van which houses the acquisition, storage and analysis equipment. Digitized data from up to 10 microphones is recorded on a Jaz disk and is analyzed post-test by microcomputer system. The RASS digitizes and stores acoustic inputs at the microphone site that can be located up to three miles from the base station and can compose a 3 mile by 3 mile array of microphones. 16-bit digitized data from the microphones is stored on removable Jaz disk and is transferred through a high speed array to a very large high speed permanent storage device. Up to 30 microphones can be utilized in the array. System control and monitoring is accomplished via Radio Frequency (RF) link. This paper will present a detailed description of both systems, along with acoustic data analysis from both systems.

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

Directory of Open Access Journals (Sweden)

Christof Stieger

2017-08-01

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

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

Science.gov (United States)

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

2010-01-01

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

Transversely Excited Multipass Photoacoustic Cell Using Electromechanical Film as Microphone

Directory of Open Access Journals (Sweden)

Jaakko Saarela

2010-05-01

Full Text Available 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.

The merit of using silicon for the development of hearing aid microphones and intraocular pressure sensors

NARCIS (Netherlands)

Bergveld, Piet

1994-01-01

An important design rule for a hearing aid is the requirement of a large signal to noise ratio, which is mainly determined by that of the microphone and its preamplifier. It will be shown that in order to increase the signal to noise ratio it is favourable to integrate the preamplifier with the

Analyzing acoustic phenomena with a smartphone microphone

Science.gov (United States)

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.

Towards Informative Path Planning for Acoustic SLAM

OpenAIRE

Evers, C; Moore, A; Naylor, P

2016-01-01

Acoustic scene mapping is a challenging task as microphone arrays can often localize sound sources only in terms of their directions. Spatial diversity can be exploited constructively to infer source-sensor range when using microphone arrays installed on moving platforms, such as robots. As the absolute location of a moving robot is often unknown in practice, Acoustic Simultaneous Localization And Mapping (a-SLAM) is required in order to localize the moving robot?s positions and jointly map t...

Array-based techniques for fingerprinting medicinal herbs

Directory of Open Access Journals (Sweden)

Xue Charlie

2011-05-01

Full Text Available Abstract Poor quality control of medicinal herbs has led to instances of toxicity, poisoning and even deaths. The fundamental step in quality control of herbal medicine is accurate identification of herbs. Array-based techniques have recently been adapted to authenticate or identify herbal plants. This article reviews the current array-based techniques, eg oligonucleotides microarrays, gene-based probe microarrays, Suppression Subtractive Hybridization (SSH-based arrays, Diversity Array Technology (DArT and Subtracted Diversity Array (SDA. We further compare these techniques according to important parameters such as markers, polymorphism rates, restriction enzymes and sample type. The applicability of the array-based methods for fingerprinting depends on the availability of genomics and genetics of the species to be fingerprinted. For the species with few genome sequence information but high polymorphism rates, SDA techniques are particularly recommended because they require less labour and lower material cost.

FILTWAM - A Framework for Online Game-based Communication Skills Training - Using Webcams and Microphones for Enhancing Learner Support

NARCIS (Netherlands)

Bahreini, Kiavash; Nadolski, Rob; Qi, Wen; Westera, Wim

2012-01-01

Bahreini, K., Nadolski, R., Qi, W., & Westera, W. (2012). FILTWAM - A Framework for Online Game-based Communication Skills Training - Using Webcams and Microphones for Enhancing Learner Support. In P. Felicia (Ed.), The 6th European Conference on Games Based Learning - ECGBL 2012 (pp. 39-48). Cork,

Biomimetic micromechanical adaptive flow-sensor arrays

Science.gov (United States)

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.

Active noise canceling system for mechanically cooled germanium radiation detectors

Science.gov (United States)

Nelson, Karl Einar; Burks, Morgan T

2014-04-22

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

Adaptive algorithm based on antenna arrays for radio communication systems

Directory of Open Access Journals (Sweden)

Fedosov Valentin

2017-01-01

Full Text Available Trends in the modern world increasingly lead to the growing popularity of wireless technologies. This is possible due to the rapid development of mobile communications, the Internet gaining high popularity, using wireless networks at enterprises, offices, buildings, etc. It requires advanced network technologies with high throughput capacity to meet the needs of users. To date, a popular destination is the development of spatial signal processing techniques allowing to increase spatial bandwidth of communication channels. The most popular method is spatial coding MIMO to increase data transmission speed which is carried out due to several spatial streams emitted by several antennas. Another advantage of this technology is the bandwidth increase to be achieved without expanding the specified frequency range. Spatial coding methods are even more attractive due to a limited frequency resource. Currently, there is an increasing use of wireless communications (for example, WiFi and WiMAX in information transmission networks. One of the main problems of evolving wireless systems is the need to increase bandwidth and improve the quality of service (reducing the error probability. Bandwidth can be increased by expanding the bandwidth or increasing the radiated power. Nevertheless, the application of these methods has some drawbacks, due to the requirements of biological protection and electromagnetic compatibility, the increase of power and the expansion of the frequency band is limited. This problem is especially relevant in mobile (cellular communication systems and wireless networks operating in difficult signal propagation conditions. One of the most effective ways to solve this problem is to use adaptive antenna arrays with weakly correlated antenna elements. Communication systems using such antennas are called MIMO systems (Multiple Input Multiple Output multiple input - multiple outputs. At the moment, existing MIMO-idea implementations do not

Extending the frequency range of free-field reciprocity calibration of measurement microphones to frequencies up to 150 kHz

DEFF Research Database (Denmark)

Barrera Figueroa, Salvador; Torras Rosell, Antoni; Jacobsen, Finn

2013-01-01

Measurement microphones are typically calibrated in a free field at frequencies up to 50 kHz. This is a sufficiently high frequency for the most sound measurement applications related with noise assessment. However, other applications such as the measurement of noise emitted by ultrasound cleanin...

Reconfigurable signal processor designs for advanced digital array radar systems

Science.gov (United States)

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

2017-05-01

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

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

Science.gov (United States)

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

2018-06-01

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

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

Science.gov (United States)

Desjardins, Jamie L

2016-01-01

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

Research status on aero-acoustic noise from wind turbine blades

International Nuclear Information System (INIS)

Yang, B

2013-01-01

This paper describes the noise mechanisms and categories of modern large wind turbine and main noise sources. Then the latest progresses in wind turbine noise researches are described from three aspects: noise prediction model, detection of noise sources by microphone array technique and methods for noise reduction. Although the turbine is restricted to horizontal axis wind turbines, the noise prediction model and reduction methods also can be applied to other turbines when the noise mechanisms are similar. Microphone array technique can be applied to locate any kind of noise sources

High Altitude Infrasound Measurements using Balloon-Borne Arrays

Science.gov (United States)

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

2015-12-01

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

Applications of the phased array technique

International Nuclear Information System (INIS)

Erhard, A.; Schenk, G.; Hauser, Th.; Voelz, U.

1999-01-01

The application of the phased array technique was limited to heavy and thick wall components as present in the nuclear industry. With the improvement of the equipment and probes other application areas are now open for the phased array technique, e.g. the inspection of the turbine blade root, weld inspection in a wall thickness range between 12 and 40 mm, inspection of aircraft components, inspection of spot welds or inspection of concretes. The aim of the use of phased array techniques has not been changed related to the first applications, i.e. the adaptation of the sound beam to the geometry by steering the angel of incidence or the skewing angle as well as the focussing of sound fields. Due to the fact, that the new applications of the phased array techniques in some cases don't leave the laboratories for the time being, the examples of this contribution will focus applications with practical background. (orig.)

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

Science.gov (United States)

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

2016-10-01

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

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

Science.gov (United States)

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

2015-12-22

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

The Fuge Tube Diode Array Spectrophotometer

Science.gov (United States)

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

2008-01-01

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

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

Science.gov (United States)

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.

A scanning bi-static SODAR

Energy Technology Data Exchange (ETDEWEB)

Behrens, P; Bradley, S [Physics Department, Auckland University, 38 Princes Street, Auckland (New Zealand); Hunerbein, S von [Acoustics Department, Newton Building, University of Salford, Greater Manchester M5 4WT (United Kingdom)

2008-05-01

Field results are given from a bi-static SODAR which uses a single central vertical transmission and three distributed microphone array receivers. Fourier transform delay methods are applied to data sampled from each microphone to retrospectively scan in angle and follow the transmitted pulse. Advantages of sampling a narrow atmospheric column, rather than distributed volumes are discussed.

A scanning bi-static SODAR

International Nuclear Information System (INIS)

Behrens, P; Bradley, S; Hunerbein, S von

2008-01-01

Field results are given from a bi-static SODAR which uses a single central vertical transmission and three distributed microphone array receivers. Fourier transform delay methods are applied to data sampled from each microphone to retrospectively scan in angle and follow the transmitted pulse. Advantages of sampling a narrow atmospheric column, rather than distributed volumes are discussed

Beamspace Adaptive Beamforming for Hydrodynamic Towed Array Self-Noise Cancellation

National Research Council Canada - National Science Library

Premus, Vincent

2001-01-01

... against signal self-nulling associated with steering vector mismatch. Particular attention is paid to the definition of white noise gain as the metric that reflects the level of mainlobe adaptive nulling for an adaptive beamformer...

Beamspace Adaptive Beamforming for Hydrodynamic Towed Array Self-Noise Cancellation

National Research Council Canada - National Science Library

Premus, Vincent

2000-01-01

... against signal self-nulling associated with steering vector mismatch. Particular attention is paid to the definition of white noise gain as the metric that reflects the level of mainlobe adaptive nulling for an adaptive beamformer...

Airframe Noise from a Hybrid Wing Body Aircraft Configuration

Science.gov (United States)

Hutcheson, Florence V.; Spalt, Taylor B.; Brooks, Thomas F.; Plassman, Gerald E.

2016-01-01

A high fidelity aeroacoustic test was conducted in the NASA Langley 14- by 22-Foot Subsonic Tunnel to establish a detailed database of component noise for a 5.8% scale HWB aircraft configuration. The model has a modular design, which includes a drooped and a stowed wing leading edge, deflectable elevons, twin verticals, and a landing gear system with geometrically scaled wheel-wells. The model is mounted inverted in the test section and noise measurements are acquired at different streamwise stations from an overhead microphone phased array and from overhead and sideline microphones. Noise source distribution maps and component noise spectra are presented for airframe configurations representing two different approach flight conditions. Array measurements performed along the aircraft flyover line show the main landing gear to be the dominant contributor to the total airframe noise, followed by the nose gear, the inboard side-edges of the LE droop, the wing tip/LE droop outboard side-edges, and the side-edges of deployed elevons. Velocity dependence and flyover directivity are presented for the main noise components. Decorrelation effects from turbulence scattering on spectral levels measured with the microphone phased array are discussed. Finally, noise directivity maps obtained from the overhead and sideline microphone measurements for the landing gear system are provided for a broad range of observer locations.

Solid state silicon based condenser microphone for hearing aid, has transducer chip and IC chip between intermediate chip and openings on both sides of intermediate chip, to allow sound towards diaphragm

DEFF Research Database (Denmark)

2000-01-01

towards diaphragm. Surface of the chip (2) has electrical conductors (14) to connect chip with IC chip (3). USE - For use in miniature electroacoustic devices such as hearing aid. ADVANTAGE - Since sound inlet is covered by filter, dust, moisture and other impurities do not obstruct interior and sound...... inlet of microphone. External electrical connection can be made economically reliable and the thermal stress is avoided with the small size solid state silicon based condenser microphone....

Adaptive RAC codes employing statistical channel evaluation ...

African Journals Online (AJOL)

An adaptive encoding technique using row and column array (RAC) codes employing a different number of parity columns that depends on the channel state is proposed in this paper. The trellises of the proposed adaptive codes and a statistical channel evaluation technique employing these trellises are designed and ...

Location of aerodynamic noise sources from a 200 kW vertical-axis wind turbine

Science.gov (United States)

Ottermo, Fredric; Möllerström, Erik; Nordborg, Anders; Hylander, Jonny; Bernhoff, Hans

2017-07-01

Noise levels emitted from a 200 kW H-rotor vertical-axis wind turbine have been measured using a microphone array at four different positions, each at a hub-height distance from the tower. The microphone array, comprising 48 microphones in a spiral pattern, allows for directional mapping of the noise sources in the range of 500 Hz to 4 kHz. The produced images indicate that most of the noise is generated in a narrow azimuth-angle range, compatible with the location where increased turbulence is known to be present in the flow, as a result of the previous passage of a blade and its support arms. It is also shown that a semi-empirical model for inflow-turbulence noise seems to produce noise levels of the correct order of magnitude, based on the amount of turbulence that could be expected from power extraction considerations.

Integration of spintronic interface for nanomagnetic arrays

Directory of Open Access Journals (Sweden)

Andrew Lyle

2011-12-01

Full Text Available An experimental demonstration utilizing a spintronic input/output (I/O interface for arrays of closely spaced nanomagnets is presented. The free layers of magnetic tunnel junctions (MTJs form dipole coupled nanomagnet arrays which can be applied to different contexts including Magnetic Quantum Cellular Automata (MQCA for logic applications and self-biased devices for field sensing applications. Dipole coupled nanomagnet arrays demonstrate adaptability to a variety of contexts due to the ability for tuning of magnetic response. Spintronics allows individual nanomagnets to be manipulated with spin transfer torque and monitored with magnetoresistance. This facilitates measurement of the magnetic coupling which is important for (yet to be demonstrated data propagation reliability studies. In addition, the same magnetic coupling can be tuned to reduce coercivity for field sensing. Dipole coupled nanomagnet arrays have the potential to be thousands of times more energy efficient than CMOS technology for logic applications, and they also have the potential to form multi-axis field sensors.

Silicon photonic micro-ring resonators to sense strain and ultrasound

NARCIS (Netherlands)

Westerveld, W.J.

2014-01-01

We demonstrated that photonic micro-ring resonators can be used in micro-machined ultrasound microphones. This might cause a breakthrough in array transducers for ultrasonography; first because optical multiplexing allows array interrogation via one optical fiber and second because the

CCD and IR array controllers

Science.gov (United States)

Leach, Robert W.; Low, Frank J.

2000-08-01

A family of controllers has bene developed that is powerful and flexible enough to operate a wide range of CCD and IR focal plane arrays in a variety of ground-based applications. These include fast readout of small CCD and IR arrays for adaptive optics applications, slow readout of large CCD and IR mosaics, and single CCD and IR array operation at low background/low noise regimes as well as high background/high speed regimes. The CCD and IR controllers have a common digital core based on user- programmable digital signal processors that are used to generate the array clocking and signal processing signals customized for each application. A fiber optic link passes image data and commands to VME or PCI interface boards resident in a host computer to the controller. CCD signal processing is done with a dual slope integrator operating at speeds of up to one Megapixel per second per channel. Signal processing of IR arrays is done either with a dual channel video processor or a four channel video processor that has built-in image memory and a coadder to 32-bit precision for operating high background arrays. Recent developments underway include the implementation of a fast fiber optic data link operating at a speed of 12.5 Megapixels per second for fast image transfer from the controller to the host computer, and supporting image acquisition software and device drivers for the PCI interface board for the Sun Solaris, Linux and Windows 2000 operating systems.

Optimization of actuator arrays for aircraft interior noise control

Science.gov (United States)

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

1993-01-01

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

First Test of Fan Active Noise Control (ANC) Completed

Science.gov (United States)

2005-01-01

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

Communication system with adaptive noise suppression

Science.gov (United States)

Kozel, David (Inventor); Devault, James A. (Inventor); Birr, Richard B. (Inventor)

2007-01-01

A signal-to-noise ratio dependent adaptive spectral subtraction process eliminates noise from noise-corrupted speech signals. The process first pre-emphasizes the frequency components of the input sound signal which contain the consonant information in human speech. Next, a signal-to-noise ratio is determined and a spectral subtraction proportion adjusted appropriately. After spectral subtraction, low amplitude signals can be squelched. A single microphone is used to obtain both the 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 unvoiced frames an estimate of the noise is obtained. A running average of the noise is used to approximate the expected value of the noise. Spectral subtraction may be performed on a composite noise-corrupted signal, or upon individual sub-bands of the noise-corrupted signal. Pre-averaging of the input signal's magnitude spectrum over multiple time frames may be performed to reduce musical noise.

METHODS FOR QUALITY ENHANCEMENT OF USER VOICE SIGNAL IN VOICE AUTHENTICATION SYSTEMS

Directory of Open Access Journals (Sweden)

O. N. Faizulaieva

2014-03-01

Full Text Available The reasonability for the usage of computer systems user voice in the authentication process is proved. The scientific task for improving the signal/noise ratio of the user voice signal in the authentication system is considered. The object of study is the process of input and output of the voice signal of authentication system user in computer systems and networks. Methods and means for input and extraction of voice signal against external interference signals are researched. Methods for quality enhancement of user voice signal in voice authentication systems are suggested. As modern computer facilities, including mobile ones, have two-channel audio card, the usage of two microphones is proposed in the voice signal input system of authentication system. Meanwhile, the task of forming a lobe of microphone array in a desired area of voice signal registration (100 Hz to 8 kHz is solved. The usage of directional properties of the proposed microphone array gives the possibility to have the influence of external interference signals two or three times less in the frequency range from 4 to 8 kHz. The possibilities for implementation of space-time processing of the recorded signals using constant and adaptive weighting factors are investigated. The simulation results of the proposed system for input and extraction of signals during digital processing of narrowband signals are presented. The proposed solutions make it possible to improve the value of the signal/noise ratio of the useful signals recorded up to 10, ..., 20 dB under the influence of external interference signals in the frequency range from 4 to 8 kHz. The results may be useful to specialists working in the field of voice recognition and speaker’s discrimination.

EXPERIMENTAL STUDY OF FIRMWARE FOR INPUT AND EXTRACTION OF USER’S VOICE SIGNAL IN VOICE AUTHENTICATION SYSTEMS

Directory of Open Access Journals (Sweden)

O. N. Faizulaieva

2014-09-01

Full Text Available Scientific task for improving the signal-to-noise ratio for user’s voice signal in computer systems and networks during the process of user’s voice authentication is considered. The object of study is the process of input and extraction of the voice signal of authentication system user in computer systems and networks. Methods and means for input and extraction of the voice signal on the background of external interference signals are investigated. Ways for quality improving of the user’s voice signal in systems of voice authentication are investigated experimentally. Firmware means for experimental unit of input and extraction of the user’s voice signal against external interference influence are considered. As modern computer means, including mobile, have two-channel audio card, two microphones are used in the voice signal input. The distance between sonic-wave sensors is 20 mm and it provides forming one direction pattern lobe of microphone array in a desired area of voice signal registration (from 100 Hz to 8 kHz. According to the results of experimental studies, the usage of directional properties of the proposed microphone array and space-time processing of the recorded signals with implementation of constant and adaptive weighting factors has made it possible to reduce considerably the influence of interference signals. The results of firmware experimental studies for input and extraction of the user’s voice signal against external interference influence are shown. The proposed solutions will give the possibility to improve the value of the signal/noise ratio of the useful signals recorded up to 20 dB under the influence of external interference signals in the frequency range from 4 to 8 kHz. The results may be useful to specialists working in the field of voice recognition and speaker discrimination.

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

Science.gov (United States)

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

2017-12-01

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

Optimization of ultrasonic arrays design and setting using a differential evolution

International Nuclear Information System (INIS)

Puel, B.; Chatillon, S.; Calmon, P.; Lesselier, D.

2011-01-01

Optimization of both design and setting of phased arrays could be not so easy when they are performed manually via parametric studies. An optimization method based on an Evolutionary Algorithm and numerical simulation is proposed and evaluated. The Randomized Adaptive Differential Evolution has been adapted to meet the specificities of the non-destructive testing applications. In particular, the solution of multi-objective problems is aimed at with the implementation of the concept of pareto-optimal sets of solutions. The algorithm has been implemented and connected to the ultrasonic simulation modules of the CIVA software used as forward model. The efficiency of the method is illustrated on two realistic cases of application: optimization of the position and delay laws of a flexible array inspecting a nozzle, considered as a mono-objective problem; and optimization of the design of a surrounded array and its delay laws, considered as a constrained bi-objective problem. (authors)

Narrowband direction of arrival estimation for antenna arrays

CERN Document Server

Foutz, Jeffrey

2008-01-01

This book provides an introduction to narrowband array signal processing, classical and subspace-based direction of arrival (DOA) estimation with an extensive discussion on adaptive direction of arrival algorithms. The book begins with a presentation of the basic theory, equations, and data models of narrowband arrays. It then discusses basic beamforming methods and describes how they relate to DOA estimation. Several of the most common classical and subspace-based direction of arrival methods are discussed. The book concludes with an introduction to subspace tracking and shows how subspace tr

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

International Nuclear Information System (INIS)

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

2011-01-01

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

Advances on Frequency Diverse Array Radar and Its Applications

Directory of Open Access Journals (Sweden)

Wang Wenqin

2018-04-01

Full Text Available Unlike the conventional phased array that provides only angle-dependent transmit beampattern, Frequency Diverse Array (FDA employs a small frequency increment across its array elements to produce automatic beam scanning without requiring phase shifters or mechanical steering. FDA can produce both rangedependent and time-variant transmit beampatterns, which overcomes the disadvantages of conventional phased arrays that produce only angle-dependent beampattern. Thus, FDA has many promising applications. Based on a previous study conducted by the author, “Frequency Diverse Array Radar: Concept, Principle and Application” (Journal of Electronics & Information Technology, 2016, 38(4: 1000–1011, the current study introduces basic FDA radar concepts, principles, and application characteristics and reviews recent advances on FDA radar and its applications. In addition, several new promising applications of FDA technology are discussed, such as radar electronic warfare and radar-communications, as well as open technical challenges such as beampattern variance, effective receiver design, adaptive signal detection and estimation, and the implementation of practical FDA radar demos.

CRISPRDetect: A flexible algorithm to define CRISPR arrays.

Science.gov (United States)

Biswas, Ambarish; Staals, Raymond H J; Morales, Sergio E; Fineran, Peter C; Brown, Chris M

2016-05-17

CRISPR (clustered regularly interspaced short palindromic repeats) RNAs provide the specificity for noncoding RNA-guided adaptive immune defence systems in prokaryotes. CRISPR arrays consist of repeat sequences separated by specific spacer sequences. CRISPR arrays have previously been identified in a large proportion of prokaryotic genomes. However, currently available detection algorithms do not utilise recently discovered features regarding CRISPR loci. We have developed a new approach to automatically detect, predict and interactively refine CRISPR arrays. It is available as a web program and command line from bioanalysis.otago.ac.nz/CRISPRDetect. CRISPRDetect discovers putative arrays, extends the array by detecting additional variant repeats, corrects the direction of arrays, refines the repeat/spacer boundaries, and annotates different types of sequence variations (e.g. insertion/deletion) in near identical repeats. Due to these features, CRISPRDetect has significant advantages when compared to existing identification tools. As well as further support for small medium and large repeats, CRISPRDetect identified a class of arrays with 'extra-large' repeats in bacteria (repeats 44-50 nt). The CRISPRDetect output is integrated with other analysis tools. Notably, the predicted spacers can be directly utilised by CRISPRTarget to predict targets. CRISPRDetect enables more accurate detection of arrays and spacers and its gff output is suitable for inclusion in genome annotation pipelines and visualisation. It has been used to analyse all complete bacterial and archaeal reference genomes.

Detection of aeroacoustic sound sources on aircraft and wind turbines

NARCIS (Netherlands)

Oerlemans, Stefan

2009-01-01

This thesis deals with the detection of aeroacoustic sound sources on aircraft and wind turbines using phased microphone arrays. First, the reliability of the array technique is assessed using airframe noise measurements in open and closed wind tunnels. It is demonstrated that quantitative acoustic

On the applicability of the spherical wave expansion with a single origin for near-field acoustical holography

DEFF Research Database (Denmark)

Gomes, J.; Hald, J.; Juhl, P.

2009-01-01

regularization (the truncated singular value decomposition) is introduced. Important differences between applying the method when using a microphone array surrounding the source completely and an array covering only a part of the source are described. Another relevant issue is the scaling of the wave functions...

Fiber optic modification of a diode array spectrophotometer

International Nuclear Information System (INIS)

Van Hare, D.R.; Prather, W.S.

1986-01-01

Fiber optics were adapted to a Hewlett-Packard diode array spectrophotometer to permit the analysis of radioactive samples without risking contamination of the instrument. Instrument performance was not compromised by the fiber optics. The instrument is in routine use at the Savannah River Plant control laboratories

Adaptive Feedforward Cancellation of Sinusoidal Disturbances in Superconducting RF Cavities

CERN Document Server

Kandil, T H; Hartung, W; Khalil, H; Popielarski, J; Vincent, J; York, R C

2004-01-01

A control method, known as adaptive feedforward cancellation (AFC) is applied to damp sinusoidal disturbances due to microphonics in superconducting RF (SRF) cavities. AFC provides a method for damping internal, and external sinusoidal disturbances with known frequencies. It is preferred over other schemes because it uses rudimentary information about the frequency response at the disturbance frequencies, without the necessity of knowing an analytic model (transfer function) of the system. It estimates the magnitude and phase of the sinusoidal disturbance inputs and generates a control signal to cancel their effect. AFC, along with a frequency estimation process, is shown to be very successful in the cancellation of sinusoidal signals from different sources. The results of this research may significantly reduce the power requirements and increase the stability for lightly loaded continuous-wave SRF systems.

Adaptation in CRISPR-Cas Systems.

Science.gov (United States)

Sternberg, Samuel H; Richter, Hagen; Charpentier, Emmanuelle; Qimron, Udi

2016-03-17

Clustered regularly interspaced short palindromic repeats (CRISPR) and CRISPR-associated (Cas) proteins constitute an adaptive immune system in prokaryotes. The system preserves memories of prior infections by integrating short segments of foreign DNA, termed spacers, into the CRISPR array in a process termed adaptation. During the past 3 years, significant progress has been made on the genetic requirements and molecular mechanisms of adaptation. Here we review these recent advances, with a focus on the experimental approaches that have been developed, the insights they generated, and a proposed mechanism for self- versus non-self-discrimination during the process of spacer selection. We further describe the regulation of adaptation and the protein players involved in this fascinating process that allows bacteria and archaea to harbor adaptive immunity. Copyright © 2016 Elsevier Inc. All rights reserved.

Design considerations for large roof-integrated photovoltaic arrays

Energy Technology Data Exchange (ETDEWEB)

Ropp, M.E.; Begovic, M.; Rohatgi, A. [Georgia Inst. of Tech., Atlanta, GA (United States); Long, R. [Georgia Institute of Technology, Atlanta (United States). Office of Facilities

1997-01-01

This paper describes calculations and modeling used in the design of the photovoltaic (PV) array built on the roof of the Georgia Tech Aquatic Center, the aquatic sports venue for the 1996 Olympic and Paralympic Games. The software package PVFORM (version 3.3) was extensively utilized; because of its importance to this work, it is thoroughly reviewed here. Procedures required to adapt PVFORM to this particular installation are described. The expected behavior and performance of the system, including maximum power output, annual energy output and maximum expected temperature, are then presented, and the use of this information in making informed design decisions is described. Finally, since the orientation of the PV array is not optimal, the effect of the unoptimized array orientation on the system`s performance is quantified. (author)

A hidden Markov model approach for determining expression from genomic tiling micro arrays

Directory of Open Access Journals (Sweden)

Krogh Anders

2006-05-01

Full Text Available Abstract Background Genomic tiling micro arrays have great potential for identifying previously undiscovered coding as well as non-coding transcription. To-date, however, analyses of these data have been performed in an ad hoc fashion. Results We present a probabilistic procedure, ExpressHMM, that adaptively models tiling data prior to predicting expression on genomic sequence. A hidden Markov model (HMM is used to model the distributions of tiling array probe scores in expressed and non-expressed regions. The HMM is trained on sets of probes mapped to regions of annotated expression and non-expression. Subsequently, prediction of transcribed fragments is made on tiled genomic sequence. The prediction is accompanied by an expression probability curve for visual inspection of the supporting evidence. We test ExpressHMM on data from the Cheng et al. (2005 tiling array experiments on ten Human chromosomes 1. Results can be downloaded and viewed from our web site 2. Conclusion The value of adaptive modelling of fluorescence scores prior to categorisation into expressed and non-expressed probes is demonstrated. Our results indicate that our adaptive approach is superior to the previous analysis in terms of nucleotide sensitivity and transfrag specificity.

Networked Airborne Communications Using Adaptive Multi Beam Directional Links

Science.gov (United States)

2016-03-05

Networked Airborne Communications Using Adaptive Multi-Beam Directional Links R. Bruce MacLeod Member, IEEE, and Adam Margetts Member, IEEE MIT...provide new techniques for increasing throughput in airborne adaptive directional net- works. By adaptive directional linking, we mean systems that can...techniques can dramatically increase the capacity in airborne networks. Advances in digital array technology are beginning to put these gains within reach

Optimized Adaptive Perturb and Observe Maximum Power Point Tracking Control for Photovoltaic Generation

Directory of Open Access Journals (Sweden)

Luigi Piegari

2015-04-01

Full Text Available The power extracted from PV arrays is usually maximized using maximum power point tracking algorithms. One of the most widely used techniques is the perturb & observe algorithm, which periodically perturbs the operating point of the PV array, sometime with an adaptive perturbation step, and compares the PV power before and after the perturbation. This paper analyses the most suitable perturbation step to optimize maximum power point tracking performance and suggests a design criterion to select the parameters of the controller. Using this proposed adaptive step, the MPPT perturb & observe algorithm achieves an excellent dynamic response by adapting the perturbation step to the actual operating conditions of the PV array. The proposed algorithm has been validated and tested in a laboratory using a dual input inductor push-pull converter. This particular converter topology is an efficient interface to boost the low voltage of PV arrays and effectively control the power flow when input or output voltages are variable. The experimental results have proved the superiority of the proposed algorithm in comparison of traditional perturb & observe and incremental conductance techniques.

Noise Quantification with Beamforming Deconvolution: Effects of Regularization and Boundary Conditions

DEFF Research Database (Denmark)

Lylloff, Oliver Ackermann; Fernandez Grande, Efren

Delay-and-sum (DAS) beamforming can be described as a linear convolution of an unknown sound source distribution and the microphone array response to a point source, i.e., point-spread function. Deconvolution tries to compensate for the influence of the array response and reveal the true source...

Speech Acquisition and Automatic Speech Recognition for Integrated Spacesuit Audio Systems

Science.gov (United States)

Huang, Yiteng; Chen, Jingdong; Chen, Shaoyan

2010-01-01

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

A practical implementation of microphone free-field comparison calibration according to the standard IEC 61094-8

DEFF Research Database (Denmark)

Barrera Figueroa, Salvador; Torras Rosell, Antoni; Rasmussen, Knud

2012-01-01

. The two methodologies assume that the two microphones are exposed to the same sound pressure. This can be achieved by measuring the ratio of output voltages either sequentially or simultaneously. The first method requires a stable source to ensure that the sound pressure is approximately the same when....... A third method, consisting of a combination of the sequential and simultaneous methodologies, has also been investigated. Though the application of time selective techniques is not discussed, the experimental results indicate the immunity to unwanted reflections in the sequential and combined approaches...... while it may be necessary to apply these techniques in the simultaneous approach....

Array capabilities and future arrays

International Nuclear Information System (INIS)

Radford, D.

1993-01-01

Early results from the new third-generation instruments GAMMASPHERE and EUROGAM are confirming the expectation that such arrays will have a revolutionary effect on the field of high-spin nuclear structure. When completed, GAMMASHPERE will have a resolving power am order of magnitude greater that of the best second-generation arrays. When combined with other instruments such as particle-detector arrays and fragment mass analysers, the capabilites of the arrays for the study of more exotic nuclei will be further enhanced. In order to better understand the limitations of these instruments, and to design improved future detector systems, it is important to have some intelligible and reliable calculation for the relative resolving power of different instrument designs. The derivation of such a figure of merit will be briefly presented, and the relative sensitivities of arrays currently proposed or under construction presented. The design of TRIGAM, a new third-generation array proposed for Chalk River, will also be discussed. It is instructive to consider how far arrays of Compton-suppressed Ge detectors could be taken. For example, it will be shown that an idealised open-quote perfectclose quotes third-generation array of 1000 detectors has a sensitivity an order of magnitude higher again than that of GAMMASPHERE. Less conventional options for new arrays will also be explored

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

Science.gov (United States)

Stephens, David

2011-01-01

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

Adaptive security systems -- Combining expert systems with adaptive technologies

International Nuclear Information System (INIS)

Argo, P.; Loveland, R.; Anderson, K.

1997-01-01

The Adaptive Multisensor Integrated Security System (AMISS) uses a variety of computational intelligence techniques to reason from raw sensor data through an array of processing layers to arrive at an assessment for alarm/alert conditions based on human behavior within a secure facility. In this paper, the authors give an overview of the system and briefly describe some of the major components of the system. This system is currently under development and testing in a realistic facility setting

Uniform Circular Antenna Array Applications in Coded DS-CDMA Mobile Communication Systems

National Research Council Canada - National Science Library

Seow, Tian

2003-01-01

...) has greatly increased. This thesis examines the use of an equally spaced circular adaptive antenna array at the mobile station for a typical coded direct sequence code division multiple access (DS-CDMA...

Adaptive Residual Interpolation for Color and Multispectral Image Demosaicking.

Science.gov (United States)

Monno, Yusuke; Kiku, Daisuke; Tanaka, Masayuki; Okutomi, Masatoshi

2017-12-01

Color image demosaicking for the Bayer color filter array is an essential image processing operation for acquiring high-quality color images. Recently, residual interpolation (RI)-based algorithms have demonstrated superior demosaicking performance over conventional color difference interpolation-based algorithms. In this paper, we propose adaptive residual interpolation (ARI) that improves existing RI-based algorithms by adaptively combining two RI-based algorithms and selecting a suitable iteration number at each pixel. These are performed based on a unified criterion that evaluates the validity of an RI-based algorithm. Experimental comparisons using standard color image datasets demonstrate that ARI can improve existing RI-based algorithms by more than 0.6 dB in the color peak signal-to-noise ratio and can outperform state-of-the-art algorithms based on training images. We further extend ARI for a multispectral filter array, in which more than three spectral bands are arrayed, and demonstrate that ARI can achieve state-of-the-art performance also for the task of multispectral image demosaicking.

A SOUND SOURCE LOCALIZATION TECHNIQUE TO SUPPORT SEARCH AND RESCUE IN LOUD NOISE ENVIRONMENTS

Science.gov (United States)

Yoshinaga, Hiroshi; Mizutani, Koichi; Wakatsuki, Naoto

At some sites of earthquakes and other disasters, rescuers search for people buried under rubble by listening for the sounds which they make. Thus developing a technique to localize sound sources amidst loud noise will support such search and rescue operations. In this paper, we discuss an experiment performed to test an array signal processing technique which searches for unperceivable sound in loud noise environments. Two speakers simultaneously played a noise of a generator and a voice decreased by 20 dB (= 1/100 of power) from the generator noise at an outdoor space where cicadas were making noise. The sound signal was received by a horizontally set linear microphone array 1.05 m in length and consisting of 15 microphones. The direction and the distance of the voice were computed and the sound of the voice was extracted and played back as an audible sound by array signal processing.

Modelling clustering of vertically aligned carbon nanotube arrays.

Science.gov (United States)

Schaber, Clemens F; Filippov, Alexander E; Heinlein, Thorsten; Schneider, Jörg J; Gorb, Stanislav N

2015-08-06

Previous research demonstrated that arrays of vertically aligned carbon nanotubes (VACNTs) exhibit strong frictional properties. Experiments indicated a strong decrease of the friction coefficient from the first to the second sliding cycle in repetitive measurements on the same VACNT spot, but stable values in consecutive cycles. VACNTs form clusters under shear applied during friction tests, and self-organization stabilizes the mechanical properties of the arrays. With increasing load in the range between 300 µN and 4 mN applied normally to the array surface during friction tests the size of the clusters increases, while the coefficient of friction decreases. To better understand the experimentally obtained results, we formulated and numerically studied a minimalistic model, which reproduces the main features of the system with a minimum of adjustable parameters. We calculate the van der Waals forces between the spherical friction probe and bunches of the arrays using the well-known Morse potential function to predict the number of clusters, their size, instantaneous and mean friction forces and the behaviour of the VACNTs during consecutive sliding cycles and at different normal loads. The data obtained by the model calculations coincide very well with the experimental data and can help in adapting VACNT arrays for biomimetic applications.

Final report on COOMET.AUV.A-S1: Technical report on supplementary comparison 'Comparison of national standards of the sound pressure unit in air through calibration of working reference microphones'

Science.gov (United States)

Pozdeeva, Valentina; Chalyy, Vladimir

2014-01-01

The supplementary comparison COOMET.AUV.A-S1 for secondary calibration methods using WS1 and WS2 measurement microphones was carried out from 2009 to 2010. The results were submitted to and approved by CCAUV in April 2014. Four National Metrology Institutes took part in this comparison and are as follows: BelGIM (Belarus), VNIIFTRI (Russia), SMU (Slovakia) and DP NDI 'Sistema' (Ukraine). Three of the above NMIs (VNIIFTRI, SMU and DP NDI 'Sistema') had earlier participated in COOMET key comparisons and one NMI (VNIIFTRI) had also participated in CCAUV key comparisons. The Comparison Reference Values were calculated as the weighted mean values from results obtained by three institutes. The comparison results show agreement for all participants in the frequency range from 20 Hz to 12.5 kHz for WS1 microphones, and in the frequency range from 20 Hz to 16 kHz for WS2 microphones. 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).

Fiber-optical microphones and accelerometers based on polymer optical fiber Bragg gratings

DEFF Research Database (Denmark)

Yuan, Scott Wu; Stefani, Alessio; Bang, Ole

2010-01-01

Polymer optical fibers (POFs) are ideal for applications as the sensing element in fiber-optical microphones and accelerometers based on fiber Bragg gratings (FBGs) due to their reduced Young’s Modulus of 3.2GPa, compared to 72GPa of Silica. To maximize the sensitivity and the dynamic range...... of the device the outer diameter and the length of the sensing fiber segment should be as small as possible. To this end we have fabricated 3mm FBGs in single-mode step-index POFs of diameter 115 micron, using 325nm UV writing and a phase-mask technique. 6mm POF sections with FBGs in the center have been glued...... to standard Silica SMF28 fibers. These POF FBGs have been characterized in terms of temperature and strain to find operating regimes with no hysteresis. Commercial fast wavelength interrogators (KHz) are shown to be able to track the thin POF FBGs and they are finally applied in a prototype accelerometer...

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

Directory of Open Access Journals (Sweden)

Yongmeng Zhang

2015-01-01

Full Text Available 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.

DUAL POLARIZATION ANTENNA ARRAY WITH VERY LOW CROSS POLARIZATION AND LOW SIDE LOBES

DEFF Research Database (Denmark)

1997-01-01

The present invention relates to an antenna array adapted to radiate or receive electromagnetic waves of one or two polarizations with very low cross polarization and low side lobes. An antenna array comprising many antenna elements, e.g. more than ten antenna elements, is provided in which...... formation of grating lobes are inhibited in selected directions of the radiation and cross polarization within the main lobe is suppressed at least 30 dB below the main lobe peak value. According to a preferred embodiment of the invention, the antenna elements of the antenna array comprise probe-fed patches...

Logical Qubit in a Linear Array of Semiconductor Quantum Dots

Directory of Open Access Journals (Sweden)

Cody Jones

2018-06-01

Full Text Available We design a logical qubit consisting of a linear array of quantum dots, we analyze error correction for this linear architecture, and we propose a sequence of experiments to demonstrate components of the logical qubit on near-term devices. To avoid the difficulty of fully controlling a two-dimensional array of dots, we adapt spin control and error correction to a one-dimensional line of silicon quantum dots. Control speed and efficiency are maintained via a scheme in which electron spin states are controlled globally using broadband microwave pulses for magnetic resonance, while two-qubit gates are provided by local electrical control of the exchange interaction between neighboring dots. Error correction with two-, three-, and four-qubit codes is adapted to a linear chain of qubits with nearest-neighbor gates. We estimate an error correction threshold of 10^{-4}. Furthermore, we describe a sequence of experiments to validate the methods on near-term devices starting from four coupled dots.

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

Directory of Open Access Journals (Sweden)

Mihailidis Alex

2009-07-01

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

Maximum power point tracking of partially shaded solar photovoltaic arrays

Energy Technology Data Exchange (ETDEWEB)

Roy Chowdhury, Shubhajit; Saha, Hiranmay [IC Design and Fabrication Centre, Department of Electronics and Telecommunication Engineering, Jadavpur University (India)

2010-09-15

The paper presents the simulation and hardware implementation of maximum power point (MPP) tracking of a partially shaded solar photovoltaic (PV) array using a variant of Particle Swarm Optimization known as Adaptive Perceptive Particle Swarm Optimization (APPSO). Under partially shaded conditions, the photovoltaic (PV) array characteristics get more complex with multiple maxima in the power-voltage characteristic. The paper presents an algorithmic technique to accurately track the maximum power point (MPP) of a PV array using an APPSO. The APPSO algorithm has also been validated in the current work. The proposed technique uses only one pair of sensors to control multiple PV arrays. This result in lower cost and higher accuracy of 97.7% compared to earlier obtained accuracy of 96.41% using Particle Swarm Optimization. The proposed tracking technique has been mapped onto a MSP430FG4618 microcontroller for tracking and control purposes. The whole system based on the proposed has been realized on a standard two stage power electronic system configuration. (author)

Improving Tumor Treating Fields Treatment Efficacy in Patients With Glioblastoma Using Personalized Array Layouts

International Nuclear Information System (INIS)

Wenger, Cornelia; Salvador, Ricardo; Basser, Peter J.; Miranda, Pedro C.

2016-01-01

Purpose: To investigate tumors of different size, shape, and location and the effect of varying transducer layouts on Tumor Treating Fields (TTFields) distribution in an anisotropic model. Methods and Materials: A realistic human head model was generated from MR images of 1 healthy subject. Four different virtual tumors were placed at separate locations. The transducer arrays were modeled to mimic the TTFields-delivering commercial device. For each tumor location, varying array layouts were tested. The finite element method was used to calculate the electric field distribution, taking into account tissue heterogeneity and anisotropy. Results: In all tumors, the average electric field induced by either of the 2 perpendicular array layouts exceeded the 1-V/cm therapeutic threshold value for TTFields effectiveness. Field strength within a tumor did not correlate with its size and shape but was higher in more superficial tumors. Additionally, it always increased when the array was adapted to the tumor's location. Compared with a default layout, the largest increase in field strength was 184%, and the highest average field strength induced in a tumor was 2.21 V/cm. Conclusions: These results suggest that adapting array layouts to specific tumor locations can significantly increase field strength within the tumor. Our findings support the idea of personalized treatment planning to increase TTFields efficacy for patients with GBM.

Improving Tumor Treating Fields Treatment Efficacy in Patients With Glioblastoma Using Personalized Array Layouts

Energy Technology Data Exchange (ETDEWEB)

Wenger, Cornelia, E-mail: cwenger@fc.ul.pt [Institute of Biophysics and Biomedical Engineering, Faculdade de Ciências, Universidade de Lisboa, Lisbon (Portugal); Salvador, Ricardo [Institute of Biophysics and Biomedical Engineering, Faculdade de Ciências, Universidade de Lisboa, Lisbon (Portugal); Basser, Peter J. [Section on Tissue Biophysics and Biomimetics, Eunice Kennedy Shriver National Institute of Child Health and Human Development, National Institutes of Health, Bethesda, Maryland (United States); Miranda, Pedro C. [Institute of Biophysics and Biomedical Engineering, Faculdade de Ciências, Universidade de Lisboa, Lisbon (Portugal)

2016-04-01

Purpose: To investigate tumors of different size, shape, and location and the effect of varying transducer layouts on Tumor Treating Fields (TTFields) distribution in an anisotropic model. Methods and Materials: A realistic human head model was generated from MR images of 1 healthy subject. Four different virtual tumors were placed at separate locations. The transducer arrays were modeled to mimic the TTFields-delivering commercial device. For each tumor location, varying array layouts were tested. The finite element method was used to calculate the electric field distribution, taking into account tissue heterogeneity and anisotropy. Results: In all tumors, the average electric field induced by either of the 2 perpendicular array layouts exceeded the 1-V/cm therapeutic threshold value for TTFields effectiveness. Field strength within a tumor did not correlate with its size and shape but was higher in more superficial tumors. Additionally, it always increased when the array was adapted to the tumor's location. Compared with a default layout, the largest increase in field strength was 184%, and the highest average field strength induced in a tumor was 2.21 V/cm. Conclusions: These results suggest that adapting array layouts to specific tumor locations can significantly increase field strength within the tumor. Our findings support the idea of personalized treatment planning to increase TTFields efficacy for patients with GBM.

NECTAr: New electronics for the Cherenkov Telescope Array

International Nuclear Information System (INIS)

Vorobiov, S.; Bolmont, J.; Corona, P.; Delagnes, E.; Feinstein, F.; Gascon, D.; Glicenstein, J.-F.; Naumann, C.L.; Nayman, P.; Sanuy, A.; Toussenel, F.; Vincent, P.

2011-01-01

The European astroparticle physics community aims to design and build the next generation array of Imaging Atmospheric Cherenkov Telescopes (IACTs), that will benefit from the experience of the existing H.E.S.S. and MAGIC detectors, and further expand the very-high energy astronomy domain. In order to gain an order of magnitude in sensitivity in the 10 GeV to >100TeV range, the Cherenkov Telescope Array (CTA) will employ 50-100 mirrors of various sizes equipped with 1000-4000 channels per camera, to be compared with the 6000 channels of the final H.E.S.S. array. A 3-year program, started in 2009, aims to build and test a demonstrator module of a generic CTA camera. We present here the NECTAr design of front-end electronics for the CTA, adapted to the trigger and data acquisition of a large IACTs array, with simple production and maintenance. Cost and camera performances are optimized by maximizing integration of the front-end electronics (amplifiers, fast analog samplers, ADCs) in an ASIC, achieving several GS/s and a few μs readout dead-time. We present preliminary results and extrapolated performances from Monte Carlo simulations.

NECTAr: New electronics for the Cherenkov Telescope Array

Energy Technology Data Exchange (ETDEWEB)

Vorobiov, S., E-mail: vorobiov@lpta.in2p3.f [LPTA, Universite Montpellier II and IN2P3/CNRS, Montpellier (France); Bolmont, J.; Corona, P. [LPNHE, Universite Paris VI and IN2P3/CNRS, Paris (France); Delagnes, E. [IRFU/DSM/CEA, Saclay, Gif-sur-Yvette (France); Feinstein, F. [LPTA, Universite Montpellier II and IN2P3/CNRS, Montpellier (France); Gascon, D. [ICC-UB, Universitat Barcelona, Barcelona (Spain); Glicenstein, J.-F. [IRFU/DSM/CEA, Saclay, Gif-sur-Yvette (France); Naumann, C.L.; Nayman, P. [LPNHE, Universite Paris VI and IN2P3/CNRS, Paris (France); Sanuy, A. [ICC-UB, Universitat Barcelona, Barcelona (Spain); Toussenel, F.; Vincent, P. [LPNHE, Universite Paris VI and IN2P3/CNRS, Paris (France)

2011-05-21

The European astroparticle physics community aims to design and build the next generation array of Imaging Atmospheric Cherenkov Telescopes (IACTs), that will benefit from the experience of the existing H.E.S.S. and MAGIC detectors, and further expand the very-high energy astronomy domain. In order to gain an order of magnitude in sensitivity in the 10 GeV to >100TeV range, the Cherenkov Telescope Array (CTA) will employ 50-100 mirrors of various sizes equipped with 1000-4000 channels per camera, to be compared with the 6000 channels of the final H.E.S.S. array. A 3-year program, started in 2009, aims to build and test a demonstrator module of a generic CTA camera. We present here the NECTAr design of front-end electronics for the CTA, adapted to the trigger and data acquisition of a large IACTs array, with simple production and maintenance. Cost and camera performances are optimized by maximizing integration of the front-end electronics (amplifiers, fast analog samplers, ADCs) in an ASIC, achieving several GS/s and a few {mu}s readout dead-time. We present preliminary results and extrapolated performances from Monte Carlo simulations.

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

OpenAIRE

Zhang, Ming

2012-01-01

Low-frequency otoacoustic emissions (OAEs) are often concealed by acoustic background noise such as those from a patient’s breathing and from the environment during recording in clinics. When using electrocochleaography (ECochG or ECoG), such as cochlear microphonics (CMs), acoustic background noise do not contaminate the recordings. Our objective is to study the response pattern of CM waveforms (CMWs) to explore an alternative approach in assessing cochlear functions. In response to a 14-mse...

Estimation of Road Vehicle Speed Using Two Omnidirectional Microphones: A Maximum Likelihood Approach

Directory of Open Access Journals (Sweden)

López-Valcarce Roberto

2004-01-01

Full Text Available We address the problem of estimating the speed of a road vehicle from its acoustic signature, recorded by a pair of omnidirectional microphones located next to the road. This choice of sensors is motivated by their nonintrusive nature as well as low installation and maintenance costs. A novel estimation technique is proposed, which is based on the maximum likelihood principle. It directly estimates car speed without any assumptions on the acoustic signal emitted by the vehicle. This has the advantages of bypassing troublesome intermediate delay estimation steps as well as eliminating the need for an accurate yet general enough acoustic traffic model. An analysis of the estimate for narrowband and broadband sources is provided and verified with computer simulations. The estimation algorithm uses a bank of modified crosscorrelators and therefore it is well suited to DSP implementation, performing well with preliminary field data.

System Realization of Broad Band Digital Beam Forming for Digital Array Radar

Directory of Open Access Journals (Sweden)

Wang Feng

2013-09-01

Full Text Available Broad band Digital Beam Forming (DBF is the key technique for the realization of Digital Array Radar (DAR. We propose the method of combination realization of the channel equalization and DBF time delay filter function by using adaptive Sample Matrix Inversion algorithm. The broad band DBF function is realized on a new DBF module based on parallel fiber optic engines and Field Program Gate Array (FPGA. Good performance is achieved when it is used to some radar products.

Comparison of candidate solar array maximum power utilization approaches. [for spacecraft propulsion

Science.gov (United States)

Costogue, E. N.; Lindena, S.

1976-01-01

A study was made of five potential approaches that can be utilized to detect the maximum power point of a solar array while sustaining operations at or near maximum power and without endangering stability or causing array voltage collapse. The approaches studied included: (1) dynamic impedance comparator, (2) reference array measurement, (3) onset of solar array voltage collapse detection, (4) parallel tracker, and (5) direct measurement. The study analyzed the feasibility and adaptability of these approaches to a future solar electric propulsion (SEP) mission, and, specifically, to a comet rendezvous mission. Such missions presented the most challenging requirements to a spacecraft power subsystem in terms of power management over large solar intensity ranges of 1.0 to 3.5 AU. The dynamic impedance approach was found to have the highest figure of merit, and the reference array approach followed closely behind. The results are applicable to terrestrial solar power systems as well as to other than SEP space missions.

Practical guidelines for implementing adaptive optics in fluorescence microscopy

Science.gov (United States)

Wilding, Dean; Pozzi, Paolo; Soloviev, Oleg; Vdovin, Gleb; Verhaegen, Michel

2018-02-01

In life sciences, interest in the microscopic imaging of increasingly complex three dimensional samples, such as cell spheroids, zebrafish embryos, and in vivo applications in small animals, is growing quickly. Due to the increasing complexity of samples, more and more life scientists are considering the implementation of adaptive optics in their experimental setups. While several approaches to adaptive optics in microscopy have been reported, it is often difficult and confusing for the microscopist to choose from the array of techniques and equipment. In this poster presentation we offer a small guide to adaptive optics providing general guidelines for successful adaptive optics implementation.

A diversified portfolio model of adaptability.

Science.gov (United States)

Chandra, Siddharth; Leong, Frederick T L

2016-12-01

A new model of adaptability, the diversified portfolio model (DPM) of adaptability, is introduced. In the 1950s, Markowitz developed the financial portfolio model by demonstrating that investors could optimize the ratio of risk and return on their portfolios through risk diversification. The DPM integrates attractive features of a variety of models of adaptability, including Linville's self-complexity model, the risk and resilience model, and Bandura's social cognitive theory. The DPM draws on the concept of portfolio diversification, positing that diversified investment in multiple life experiences, life roles, and relationships promotes positive adaptation to life's challenges. The DPM provides a new integrative model of adaptability across the biopsychosocial levels of functioning. More importantly, the DPM addresses a gap in the literature by illuminating the antecedents of adaptive processes studied in a broad array of psychological models. The DPM is described in relation to the biopsychosocial model and propositions are offered regarding its utility in increasing adaptiveness. Recommendations for future research are also offered. (PsycINFO Database Record (c) 2016 APA, all rights reserved).

Projection neuron circuits resolved using correlative array tomography

Directory of Open Access Journals (Sweden)

Daniele eOberti

2011-04-01

Full Text Available Assessment of three-dimensional morphological structure and synaptic connectivity is essential for a comprehensive understanding of neural processes controlling behavior. Different microscopy approaches have been proposed based on light microcopy (LM, electron microscopy (EM, or a combination of both. Correlative array tomography (CAT is a technique in which arrays of ultrathin serial sections are repeatedly stained with fluorescent antibodies against synaptic molecules and neurotransmitters and imaged with LM and EM (Micheva and Smith, 2007. The utility of this correlative approach is limited by the ability to preserve fluorescence and antigenicity on the one hand, and EM tissue ultrastructure on the other. We demonstrate tissue staining and fixation protocols and a workflow that yield an excellent compromise between these multimodal imaging constraints. We adapt CAT for the study of projection neurons between different vocal brain regions in the songbird. We inject fluorescent tracers of different colors into afferent and efferent areas of HVC in zebra finches. Fluorescence of some tracers is lost during tissue preparation but recovered using anti-dye antibodies. Synapses are identified in EM imagery based on their morphology and ultrastructure and classified into projection neuron type based on fluorescence signal. Our adaptation of array tomography, involving the use of fluorescent tracers and heavy-metal rich staining and embedding protocols for high membrane contrast in EM will be useful for research aimed at statistically describing connectivity between different projection neuron types and for elucidating how sensory signals are routed in the brain and transformed into a meaningful motor output.

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

Directory of Open Access Journals (Sweden)

Tobias eNeher

2014-12-01

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

Micro-phonics analysis and compensation with a feedback loop at low cavity gradient

International Nuclear Information System (INIS)

Luong, M.; Devanz, G.; Jacques, E.; Novo, J.; Neumann, A.; Kugeler, O.

2007-10-01

For FEL projects based on a superconducting linac operating in continuous wave (CW) mode, the RF power optimization finally comes up against the micro-phonics disturbances, which result in an unpredictable detuning of the cavities. A new piezoelectric tuner was developed and mounted on a TTF 9-cells cavity with an appropriate instrumentation. This system enables a full characterization of the disturbances and the tuner behavior. The experimental results pointed out 3 distinct regimes of perturbation: a strong but quickly damped very low frequency oscillation due to cryogenic control, a quasi-stationary oscillation around 50 and 100 Hz due to the operation of vacuum pumps motor and some lower amplitudes oscillations related to the excitation of the mechanical structure Eigenmodes by environmental noise. Modeling, simulations and experimental validations were carried out to demonstrate the feasibility of a feedback compensation for a multi-cell cavity. The results also bring to some recommendations that may overcome the limitations pointed out in the present experimentation

A real-time regional adaptive exposure method for saving dose-area product in x-ray fluoroscopy

International Nuclear Information System (INIS)

Burion, Steve; Funk, Tobias; Speidel, Michael A.

2013-01-01

Purpose: Reduction of radiation dose in x-ray imaging has been recognized as a high priority in the medical community. Here the authors show that a regional adaptive exposure method can reduce dose-area product (DAP) in x-ray fluoroscopy. The authors' method is particularly geared toward providing dose savings for the pediatric population. Methods: The scanning beam digital x-ray system uses a large-area x-ray source with 8000 focal spots in combination with a small photon-counting detector. An imaging frame is obtained by acquiring and reconstructing up to 8000 detector images, each viewing only a small portion of the patient. Regional adaptive exposure was implemented by varying the exposure of the detector images depending on the local opacity of the object. A family of phantoms ranging in size from infant to obese adult was imaged in anteroposterior view with and without adaptive exposure. The DAP delivered to each phantom was measured in each case, and noise performance was compared by generating noise arrays to represent regional noise in the images. These noise arrays were generated by dividing the image into regions of about 6 mm 2 , calculating the relative noise in each region, and placing the relative noise value of each region in a one-dimensional array (noise array) sorted from highest to lowest. Dose-area product savings were calculated as the difference between the ratio of DAP with adaptive exposure to DAP without adaptive exposure. The authors modified this value by a correction factor that matches the noise arrays where relative noise is the highest to report a final dose-area product savings. Results: The average dose-area product saving across the phantom family was (42 ± 8)% with the highest dose-area product saving in the child-sized phantom (50%) and the lowest in the phantom mimicking an obese adult (23%). Conclusions: Phantom measurements indicate that a regional adaptive exposure method can produce large DAP savings without compromising

Active Micro structured Optical Arrays of Grazing Incidence Reflectors

International Nuclear Information System (INIS)

Willingale, R.; Feldman, Ch.; Michette, A.; Hart, D.; McFaul, Ch; Morrison, G.R.; Pfauntsch, S.; Powell, A.K.; Sahraei, Sh.; Shand, M.T.; Button, T.; Rodriguez-Sanmartin, D.; Zhang, D.; Dunare, C.; Parkes, W.; Stevenson, T.; Folkard, M.; Vojnovic, B.; Vojnovic, B.

2011-01-01

The UK Smart X-Ray Optics (SXO) programme is developing active/adaptive optics for terrestrial applications. One of the technologies proposed is micro structured optical arrays (MOAs), which focus X-rays using grazing incidence reflection through consecutive aligned arrays of microscopic channels. Although such arrays are similar in concept to poly capillary and microchannel plate optics, they can be bent and adjusted using piezoelectric actuators providing control over the focusing and inherent aberrations. Custom configurations can be designed, using ray tracing and finite element analysis, for applications from sub-keV to several-keV X-rays, and the channels of appropriate aspect ratios can be made using deep silicon etching. An exemplar application will be in the micro probing of biological cells and tissue samples using Ti Ka radiation (4.5?keV) in studies related to radiation-induced cancers. This paper discusses the optical design, modelling, and manufacture of such optics

Adaptive Space-Time, Processing for High Performance, Robust Military Wireless Communications

National Research Council Canada - National Science Library

Haimovich, Alexander

2000-01-01

...: (I) performance of adaptive arrays for wireless communications over fading channels in the presence of cochannel interference particularly the case when the number of interference sources exceeds...

3D-SoftChip: A Novel Architecture for Next-Generation Adaptive Computing Systems

Directory of Open Access Journals (Sweden)

Lee Mike Myung-Ok

2006-01-01

Full Text Available This paper introduces a novel architecture for next-generation adaptive computing systems, which we term 3D-SoftChip. The 3D-SoftChip is a 3-dimensional (3D vertically integrated adaptive computing system combining state-of-the-art processing and 3D interconnection technology. It comprises the vertical integration of two chips (a configurable array processor and an intelligent configurable switch through an indium bump interconnection array (IBIA. The configurable array processor (CAP is an array of heterogeneous processing elements (PEs, while the intelligent configurable switch (ICS comprises a switch block, 32-bit dedicated RISC processor for control, on-chip program/data memory, data frame buffer, along with a direct memory access (DMA controller. This paper introduces the novel 3D-SoftChip architecture for real-time communication and multimedia signal processing as a next-generation computing system. The paper further describes the advanced HW/SW codesign and verification methodology, including high-level system modeling of the 3D-SoftChip using SystemC, being used to determine the optimum hardware specification in the early design stage.

Capturing and reproducing realistic acoustic scenes for hearing research

DEFF Research Database (Denmark)

Marschall, Marton; Buchholz, Jörg

Accurate spatial audio recordings are important for a range of applications, from the creation of realistic virtual sound environments to the evaluation of communication devices, such as hearing instruments and mobile phones. Spherical microphone arrays are particularly well-suited for capturing....... The properties of MOA microphone layouts and processing were investigated further by considering several order combinations. It was shown that the performance for horizontal vs. elevated sources can be adjusted by varying the order combination, but that a benefit of the higher horizontal orders can only be seen...

New measurements techniques

DEFF Research Database (Denmark)

Torras Rosell, Antoni

(NAH), the suggested holographic method features novel spectral properties in the wavenumber domain. On the other hand, an acousto-optic beamformer has been designed and validated experimentally for the localization of sound sources located in the far field. In this case, a laser beam is interpreted...... as a line array of microphones with infinite resolution, which makes the proposed acousto-optic beamformer immune to spatial aliasing. In addition, the present PhD study investigates the applicability of photon correlation spectroscopy as a primary method for microphone calibration under free...

Adaptive transmit selection with interference suppression

KAUST Repository

Radaydeh, Redha Mahmoud Mesleh

2010-01-01

This paper studies the performance of adaptive transmit channel selection in multipath fading channels. The adaptive selection algorithms are configured for single-antenna bandwidth-efficient or power-efficient transmission with as low transmit channel estimations as possible. Due to the fact that the number of active co-channel interfering signals and their corresponding powers experience random behavior, the adaptation to channels conditions, assuming uniform buffer and traffic loading, is proposed to be jointly based on the transmit channels instantaneous signal-to-noise ratios (SNRs) and signal-to- interference-plus- noise ratios (SINRs). Two interference cancelation algorithms, which are the dominant cancelation and the less complex arbitrary cancelation, are considered, for which the receive antenna array is assumed to have small angular spread. Analytical formulation for some performance measures in addition to several processing complexity and numerical comparisons between various adaptation schemes are presented. ©2010 IEEE.

Micro-magnet arrays for specific single bacterial cell positioning

Energy Technology Data Exchange (ETDEWEB)

Pivetal, Jérémy, E-mail: jeremy.piv@netcmail.com [Ecole Centrale de Lyon, CNRS UMR 5005, Laboratoire Ampère, F-69134 Écully (France); Royet, David [Ecole Centrale de Lyon, CNRS UMR 5005, Laboratoire Ampère, F-69134 Écully (France); Ciuta, Georgeta [Univ. Grenoble Alpes, Inst NEEL, F-38042 Grenoble (France); CNRS, Inst NEEL, F-38042 Grenoble (France); Frenea-Robin, Marie [Université de Lyon, Université Lyon 1, CNRS UMR 5005, Laboratoire Ampère, F-69622 Villeurbanne (France); Haddour, Naoufel [Ecole Centrale de Lyon, CNRS UMR 5005, Laboratoire Ampère, F-69134 Écully (France); Dempsey, Nora M. [Univ. Grenoble Alpes, Inst NEEL, F-38042 Grenoble (France); CNRS, Inst NEEL, F-38042 Grenoble (France); Dumas-Bouchiat, Frédéric [Univ Limoges, CNRS, SPCTS UMR 7513, 12 Rue Atlantis, F-87068 Limoges (France); Simonet, Pascal [Ecole Centrale de Lyon, CNRS UMR 5005, Laboratoire Ampère, F-69134 Écully (France)

2015-04-15

In various contexts such as pathogen detection or analysis of microbial diversity where cellular heterogeneity must be taken into account, there is a growing need for tools and methods that enable microbiologists to analyze bacterial cells individually. One of the main challenges in the development of new platforms for single cell studies is to perform precise cell positioning, but the ability to specifically target cells is also important in many applications. In this work, we report the development of new strategies to selectively trap single bacterial cells upon large arrays, based on the use of micro-magnets. Escherichia coli bacteria were used to demonstrate magnetically driven bacterial cell organization. In order to provide a flexible approach adaptable to several applications in the field of microbiology, cells were magnetically and specifically labeled using two different strategies, namely immunomagnetic labeling and magnetic in situ hybridization. Results show that centimeter-sized arrays of targeted, isolated bacteria can be successfully created upon the surface of a flat magnetically patterned hard magnetic film. Efforts are now being directed towards the integration of a detection tool to provide a complete micro-system device for a variety of microbiological applications. - Highlights: 1.We report a new approach to selectively micropattern bacterial cells individually upon micro-magnet arrays. 2.Permanent micro-magnets of a size approaching that of bacteria could be fabricated using a Thermo-Magnetic Patterning process. 3.Bacterial cells were labeled using two different magnetic labeling strategies providing flexible approach adaptable to several applications in the field of microbiology.

Improvement of detection of stress corrosion cracks with ultrasonic phased array probes

International Nuclear Information System (INIS)

Wustenberg, H.; Mohrle, W.; Wegner, W.; Schenk, G.; Erhard, A.

1986-01-01

Probes with linear arrays can be used for the detection of stress corrosion cracks especially if the variability of the sound field is used to change the skewing angle of angle beam probes. The phased array concept can be used to produce a variable skewing angle or a variable angle of incidence depending on the orientation of the linear array on the wedge. This helps to adapt the direction of the ultrasonic beam to probable crack orientations. It has been demonstrated with artificial reflectors as well as with corrosion cracks, that the detection of misoriented cracks can be improved by this approach. The experiences gained during the investigations are encouraging the application of phased array probes for stress corrosion phenomena close to the heat effected zone of welds. Probes with variable skewing angles may find some interesting applications on welds in tubular structures e.g., at off shore constructions and on some difficult geometries within the primary circuit of nuclear power plants

Focal plane array with modular pixel array components for scalability

Science.gov (United States)

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

2014-12-09

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

Unvoiced Speech Recognition Using Tissue-Conductive Acoustic Sensor

Directory of Open Access Journals (Sweden)

Heracleous Panikos

2007-01-01

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

Copper-encapsulated vertically aligned carbon nanotube arrays.

Science.gov (United States)

Stano, Kelly L; Chapla, Rachel; Carroll, Murphy; Nowak, Joshua; McCord, Marian; Bradford, Philip D

2013-11-13

A new procedure is described for the fabrication of vertically aligned carbon nanotubes (VACNTs) that are decorated, and even completely encapsulated, by a dense network of copper nanoparticles. The process involves the conformal deposition of pyrolytic carbon (Py-C) to stabilize the aligned carbon-nanotube structure during processing. The stabilized arrays are mildly functionalized using oxygen plasma treatment to improve wettability, and they are then infiltrated with an aqueous, supersaturated Cu salt solution. Once dried, the salt forms a stabilizing crystal network throughout the array. After calcination and H2 reduction, Cu nanoparticles are left decorating the CNT surfaces. Studies were carried out to determine the optimal processing parameters to maximize Cu content in the composite. These included the duration of Py-C deposition and system process pressure as well as the implementation of subsequent and multiple Cu salt solution infiltrations. The optimized procedure yielded a nanoscale hybrid material where the anisotropic alignment from the VACNT array was preserved, and the mass of the stabilized arrays was increased by over 24-fold because of the addition of Cu. The procedure has been adapted for other Cu salts and can also be used for other metal salts altogether, including Ni, Co, Fe, and Ag. The resulting composite is ideally suited for application in thermal management devices because of its low density, mechanical integrity, and potentially high thermal conductivity. Additionally, further processing of the material via pressing and sintering can yield consolidated, dense bulk composites.

Springer handbook of acoustics

CERN Document Server

2014-01-01

Acoustics, the science of sound, has developed into a broad interdisciplinary field encompassing the academic disciplines of physics, engineering, psychology, speech, audiology, music, architecture, physiology, neuroscience, and electronics. The Springer Handbook of Acoustics is also in his 2nd edition an unparalleled modern handbook reflecting this richly interdisciplinary nature edited by one of the acknowledged masters in the field, Thomas Rossing. Researchers and students benefit from the comprehensive contents. This new edition of the Handbook features over 11 revised and expanded chapters, new illustrations, and 2 new chapters covering microphone arrays  and acoustic emission.  Updated chapters contain the latest research and applications in, e.g. sound propagation in the atmosphere, nonlinear acoustics in fluids, building and concert hall acoustics, signal processing, psychoacoustics, computer music, animal bioacousics, sound intensity, modal acoustics as well as new chapters on microphone arrays an...

ATMAD: robust image analysis for Automatic Tissue MicroArray De-arraying.

Science.gov (United States)

Nguyen, Hoai Nam; Paveau, Vincent; Cauchois, Cyril; Kervrann, Charles

2018-04-19

Over the last two decades, an innovative technology called Tissue Microarray (TMA), which combines multi-tissue and DNA microarray concepts, has been widely used in the field of histology. It consists of a collection of several (up to 1000 or more) tissue samples that are assembled onto a single support - typically a glass slide - according to a design grid (array) layout, in order to allow multiplex analysis by treating numerous samples under identical and standardized conditions. However, during the TMA manufacturing process, the sample positions can be highly distorted from the design grid due to the imprecision when assembling tissue samples and the deformation of the embedding waxes. Consequently, these distortions may lead to severe errors of (histological) assay results when the sample identities are mismatched between the design and its manufactured output. The development of a robust method for de-arraying TMA, which localizes and matches TMA samples with their design grid, is therefore crucial to overcome the bottleneck of this prominent technology. In this paper, we propose an Automatic, fast and robust TMA De-arraying (ATMAD) approach dedicated to images acquired with brightfield and fluorescence microscopes (or scanners). First, tissue samples are localized in the large image by applying a locally adaptive thresholding on the isotropic wavelet transform of the input TMA image. To reduce false detections, a parametric shape model is considered for segmenting ellipse-shaped objects at each detected position. Segmented objects that do not meet the size and the roundness criteria are discarded from the list of tissue samples before being matched with the design grid. Sample matching is performed by estimating the TMA grid deformation under the thin-plate model. Finally, thanks to the estimated deformation, the true tissue samples that were preliminary rejected in the early image processing step are recognized by running a second segmentation step. We

Timed arrays wideband and time varying antenna arrays

CERN Document Server

Haupt, Randy L

2015-01-01

Introduces timed arrays and design approaches to meet the new high performance standards The author concentrates on any aspect of an antenna array that must be viewed from a time perspective. The first chapters briefly introduce antenna arrays and explain the difference between phased and timed arrays. Since timed arrays are designed for realistic time-varying signals and scenarios, the book also reviews wideband signals, baseband and passband RF signals, polarization and signal bandwidth. Other topics covered include time domain, mutual coupling, wideband elements, and dispersion. The auth

Developing a gate-array capability at a research and development laboratory

Science.gov (United States)

Balch, J. W.; Current, K. W.; Magnuson, W. G., Jr.; Pocha, M. D.

1983-03-01

Experiences in developing a gate array capability for low volume applications in a research and development (R and D) laboratory are described. By purchasing unfinished wafers and doing the customization steps in-house. Turnaround time was shortened to as little as one week and the direct costs reduced to as low as $5K per design. Designs generally require fast turnaround (a few weeks to a few months) and very low volumes (1 to 25). Design costs must be kept at a minimum. After reviewing available commercial gate array design and fabrication services, it was determined that objectives would best be met by using existing internal integrated circuit fabrication facilities, the COMPUTERVISION interactive graphics layout system, and extensive computational capabilities. The reasons and the approach taken for; selection for a particular gate array wafer, adapting a particular logic simulation program, and how layout aids were enhanced are discussed. Testing of the customized chips is described. The content, schedule, and results of the internal gate array course recently completed are discussed. Finally, problem areas and near term plans are presented.

ArrayBridge: Interweaving declarative array processing with high-performance computing

Energy Technology Data Exchange (ETDEWEB)

Xing, Haoyuan [The Ohio State Univ., Columbus, OH (United States); Floratos, Sofoklis [The Ohio State Univ., Columbus, OH (United States); Blanas, Spyros [The Ohio State Univ., Columbus, OH (United States); Byna, Suren [Lawrence Berkeley National Lab. (LBNL), Berkeley, CA (United States); Prabhat, Prabhat [Lawrence Berkeley National Lab. (LBNL), Berkeley, CA (United States); Wu, Kesheng [Lawrence Berkeley National Lab. (LBNL), Berkeley, CA (United States); Brown, Paul [Paradigm4, Inc., Waltham, MA (United States)

2017-05-04

Scientists are increasingly turning to datacenter-scale computers to produce and analyze massive arrays. Despite decades of database research that extols the virtues of declarative query processing, scientists still write, debug and parallelize imperative HPC kernels even for the most mundane queries. This impedance mismatch has been partly attributed to the cumbersome data loading process; in response, the database community has proposed in situ mechanisms to access data in scientific file formats. Scientists, however, desire more than a passive access method that reads arrays from files. This paper describes ArrayBridge, a bi-directional array view mechanism for scientific file formats, that aims to make declarative array manipulations interoperable with imperative file-centric analyses. Our prototype implementation of ArrayBridge uses HDF5 as the underlying array storage library and seamlessly integrates into the SciDB open-source array database system. In addition to fast querying over external array objects, ArrayBridge produces arrays in the HDF5 file format just as easily as it can read from it. ArrayBridge also supports time travel queries from imperative kernels through the unmodified HDF5 API, and automatically deduplicates between array versions for space efficiency. Our extensive performance evaluation in NERSC, a large-scale scientific computing facility, shows that ArrayBridge exhibits statistically indistinguishable performance and I/O scalability to the native SciDB storage engine.

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

Directory of Open Access Journals (Sweden)

Carlos E. Galván-Tejada

2015-08-01

Full Text Available 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.

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

Science.gov (United States)

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.

A wideband wireless neural stimulation platform for high-density microelectrode arrays.

Science.gov (United States)

Myers, Frank B; Simpson, Jim A; Ghovanloo, Maysam

2006-01-01

We describe a system that allows researchers to control an implantable neural microstimulator from a PC via a USB 2.0 interface and a novel dual-carrier wireless link, which provides separate data and power transmission. Our wireless stimulator, Interestim-2B (IS-2B), is a modular device capable of generating controlled-current stimulation pulse trains across 32 sites per module with support for a variety of stimulation schemes (biphasic/monophasic, bipolar/monopolar). We have developed software to generate multi-site stimulation commands for the IS-2B based on streaming data from artificial sensory devices such as cameras and microphones. For PC interfacing, we have developed a USB 2.0 microcontroller-based interface. Data is transmitted using frequency-shift keying (FSK) at 6/12 MHz to achieve a data rate of 3 Mb/s via a pair of rectangular coils. Power is generated using a class-E power amplifier operating at 1 MHz and transmitted via a separate pair of spiral planar coils which are oriented perpendicular to the data coils to minimize cross-coupling. We have successfully demonstrated the operation of the system by applying it as a visual prosthesis. Pulse-frequency modulated stimuli are generated in real-time based on a grayscale image from a webcam. These pulses are projected onto an 11x11 LED matrix that represents a 2D microelectrode array.

Analisis Koefisien Absorpsi Bunyi Pada Komposit Penguat Serat Alam Dengan Menggunakan Alat Uji Tabung Impedansi 2 Microphone

Directory of Open Access Journals (Sweden)

Cok Istri Putri Kusuma Kencanawati

2016-12-01

Full Text Available Abstrak: Dalam perambatannya gelombang bunyi dapat di halangi dengan menggunakan suatu medium yang memiliki sifat-sifatkedap suara, sehingga energi yang ditransmisikan akan mampu dikurangi/dihambat oleh medium tersebut. Salah satumetode yang dapat dipergunakan untuk mengetahui kemampuan peredaman (koefisien absorpsi suatu mediumterhadap gelombang bunyi yang datang dapat diketahui dengan menggunakan Tabung Impedansi 2 Microphone.Sedangkan sebagai mediumnya adalah panel komposit. Mengingat dalam perkembangan ilmu bahan saat ini banyak digunakan komposit dengan penguat serat alam, dan salah satu sifat yang dikaji adalah sifat akustiknya. Kajian ini lebihmenitik beratkan sifat akustik komposit berpenguat serat alam, mengingat selama ini banyak serat alam yang terbuangpercuma menjadi limbahsehingga mencemari lingkungan. Jenis-jenis serat alam yang digunakan sebagai penguatantara lain : serat serabut kelapa, serat jerami, serat batang pisang, serat nenas, serat kapuk dan serat batang kelapasawit, sedangkan frekuensi pengukuran koefisien absorpsi terhadap medium ini berkisar anatra 200 hz sampai dengan1400 hz, dengan ketebalan spesiemn uji antara 2 mm sampai dengan 4 mmdengan menggunakan metode pengujianTabung impedansi 2 mikrophone, sesuai dengan standart ISO 10534-2:1998 and American Standart forTestingMaterials (ASTM E1050-98. Dalam kajian ini diperoleh kesimpulan bahwa pada frekuensi rendah koefisienabsorpsi bahan cukup tinggi antara 0,4 sampai dengan 0,6 dan kemampuan serap bunyi ini akan menurun denganmeningkatnya frekuensi, sedangkan pengaruh ketebalan bahan juga mempengaruhi sifat akustiknya.Kata kunci: komposit, serat alam, koefisien absorpsi, tabung impedansi Abstract: In the propagation of sound waves can be prevented by using a medium that has properties soundproofed, so that thetransmitted energy to be able to be reduced / inhibited by the medium. One method that can be used to determine theability of damping (absorption coefficient of a

Adaptively optimizing stochastic resonance in visual system

Science.gov (United States)

Yang, Tao

1998-08-01

Recent psychophysics experiment has showed that the noise strength could affect the perceived image quality. This work gives an adaptive process for achieving the optimal perceived image quality in a simple image perception array, which is a simple model of an image sensor. A reference image from memory is used for constructing a cost function and defining the optimal noise strength where the cost function gets its minimum point. The reference image is a binary image, which is used to define the background and the object. Finally, an adaptive algorithm is proposed for searching the optimal noise strength. Computer experimental results show that if the reference image is a thresholded version of the sub-threshold input image then the output of the sensor array gives an optimal output, in which the background and the object have the biggest contrast. If the reference image is different from a thresholded version of the sub-threshold input image then the output usually gives a sub-optimal contrast between the object and the background.

New customizable phased array UT instrument opens door for furthering research and better industrial implementation

International Nuclear Information System (INIS)

Dao, Gavin; Ginzel, Robert

2014-01-01

Phased array UT as an inspection technique in itself continues to gain wide acceptance. However, there is much room for improvement in terms of implementation of Phased Array (PA) technology for every unique NDT application across several industries (e.g. oil and petroleum, nuclear and power generation, steel manufacturing, etc.). Having full control of the phased array instrument and customizing a software solution is necessary for more seamless and efficient inspections, from setting the PA parameters, collecting data and reporting, to the final analysis. NDT researchers and academics also need a flexible and open platform to be able to control various aspects of the phased array process. A high performance instrument with advanced PA features, faster data rates, a smaller form factor, and capability to adapt to specific applications, will be discussed

Simulation Based Investigation of Focusing Phased Array Ultrasound in Dissimilar Metal Welds

Directory of Open Access Journals (Sweden)

Hun-Hee Kim

2016-02-01

Full Text Available Flaws at dissimilar metal welds (DMWs, such as reactor coolant systems components, Control Rod Drive Mechanism (CRDM, Bottom Mounted Instrumentation (BMI etc., in nuclear power plants have been found. Notably, primary water stress corrosion cracking (PWSCC in the DMWs could cause significant reliability problems at nuclear power plants. Therefore, phased array ultrasound is widely used for inspecting surface break cracks and stress corrosion cracks in DMWs. However, inspection of DMWs using phased array ultrasound has a relatively low probability of detection of cracks, because the crystalline structure of welds causes distortion and splitting of the ultrasonic beams which propagates anisotropic medium. Therefore, advanced evaluation techniques of phased array ultrasound are needed for improvement in the probability of detection of flaws in DMWs. Thus, in this study, an investigation of focusing and steering phased array ultrasound in DMWs was carried out using a time reversal technique, and an adaptive focusing technique based on finite element method (FEM simulation. Also, evaluation of focusing performance of three different focusing techniques was performed by comparing amplitude of phased array ultrasonic signals scattered from the targeted flaw with three different time delays.

Multicoil resonance-based parallel array for smart wireless power delivery.

Science.gov (United States)

Mirbozorgi, S A; Sawan, M; Gosselin, B

2013-01-01

This paper presents a novel resonance-based multicoil structure as a smart power surface to wirelessly power up apparatus like mobile, animal headstage, implanted devices, etc. The proposed powering system is based on a 4-coil resonance-based inductive link, the resonance coil of which is formed by an array of several paralleled coils as a smart power transmitter. The power transmitter employs simple circuit connections and includes only one power driver circuit per multicoil resonance-based array, which enables higher power transfer efficiency and power delivery to the load. The power transmitted by the driver circuit is proportional to the load seen by the individual coil in the array. Thus, the transmitted power scales with respect to the load of the electric/electronic system to power up, and does not divide equally over every parallel coils that form the array. Instead, only the loaded coils of the parallel array transmit significant part of total transmitted power to the receiver. Such adaptive behavior enables superior power, size and cost efficiency then other solutions since it does not need to use complex detection circuitry to find the location of the load. The performance of the proposed structure is verified by measurement results. Natural load detection and covering 4 times bigger area than conventional topologies with a power transfer efficiency of 55% are the novelties of presented paper.

A Real-Time Capable Software-Defined Receiver Using GPU for Adaptive Anti-Jam GPS Sensors

Science.gov (United States)

Seo, Jiwon; Chen, Yu-Hsuan; De Lorenzo, David S.; Lo, Sherman; Enge, Per; Akos, Dennis; Lee, Jiyun

2011-01-01

Due to their weak received signal power, Global Positioning System (GPS) signals are vulnerable to radio frequency interference. Adaptive beam and null steering of the gain pattern of a GPS antenna array can significantly increase the resistance of GPS sensors to signal interference and jamming. Since adaptive array processing requires intensive computational power, beamsteering GPS receivers were usually implemented using hardware such as field-programmable gate arrays (FPGAs). However, a software implementation using general-purpose processors is much more desirable because of its flexibility and cost effectiveness. This paper presents a GPS software-defined radio (SDR) with adaptive beamsteering capability for anti-jam applications. The GPS SDR design is based on an optimized desktop parallel processing architecture using a quad-core Central Processing Unit (CPU) coupled with a new generation Graphics Processing Unit (GPU) having massively parallel processors. This GPS SDR demonstrates sufficient computational capability to support a four-element antenna array and future GPS L5 signal processing in real time. After providing the details of our design and optimization schemes for future GPU-based GPS SDR developments, the jamming resistance of our GPS SDR under synthetic wideband jamming is presented. Since the GPS SDR uses commercial-off-the-shelf hardware and processors, it can be easily adopted in civil GPS applications requiring anti-jam capabilities. PMID:22164116

A Real-Time Capable Software-Defined Receiver Using GPU for Adaptive Anti-Jam GPS Sensors

Directory of Open Access Journals (Sweden)

Dennis Akos

2011-09-01

Full Text Available Due to their weak received signal power, Global Positioning System (GPS signals are vulnerable to radio frequency interference. Adaptive beam and null steering of the gain pattern of a GPS antenna array can significantly increase the resistance of GPS sensors to signal interference and jamming. Since adaptive array processing requires intensive computational power, beamsteering GPS receivers were usually implemented using hardware such as field-programmable gate arrays (FPGAs. However, a software implementation using general-purpose processors is much more desirable because of its flexibility and cost effectiveness. This paper presents a GPS software-defined radio (SDR with adaptive beamsteering capability for anti-jam applications. The GPS SDR design is based on an optimized desktop parallel processing architecture using a quad-core Central Processing Unit (CPU coupled with a new generation Graphics Processing Unit (GPU having massively parallel processors. This GPS SDR demonstrates sufficient computational capability to support a four-element antenna array and future GPS L5 signal processing in real time. After providing the details of our design and optimization schemes for future GPU-based GPS SDR developments, the jamming resistance of our GPS SDR under synthetic wideband jamming is presented. Since the GPS SDR uses commercial-off-the-shelf hardware and processors, it can be easily adopted in civil GPS applications requiring anti-jam capabilities.

Distributed Max-SINR Speech Enhancement with Ad Hoc Microphone Arrays

DEFF Research Database (Denmark)

Tavakoli, Vincent Mohammad; Jensen, Jesper Rindom; Heusdens, Richard

2017-01-01

-SINR) criterion is used with the primal-dual method of multipliers for distributed filtering. The paper investigates the convergence of the algorithm in both synchronous and asynchronous schemes, and also discusses some practical pros and cons. The applicability of the proposed method is demonstrated by means...

Integrated NEMS and optoelectronics for sensor applications.

Energy Technology Data Exchange (ETDEWEB)

Czaplewski, David A.; Serkland, Darwin Keith; Olsson, Roy H., III; Bogart, Gregory R. (Symphony Acoustics, Rio Rancho, NM); Krishnamoorthy, Uma; Warren, Mial E.; Carr, Dustin Wade (Symphony Acoustics, Rio Rancho, NM); Okandan, Murat; Peterson, Kenneth Allen

2008-01-01

This work utilized advanced engineering in several fields to find solutions to the challenges presented by the integration of MEMS/NEMS with optoelectronics to realize a compact sensor system, comprised of a microfabricated sensor, VCSEL, and photodiode. By utilizing microfabrication techniques in the realization of the MEMS/NEMS component, the VCSEL and the photodiode, the system would be small in size and require less power than a macro-sized component. The work focused on two technologies, accelerometers and microphones, leveraged from other LDRD programs. The first technology was the nano-g accelerometer using a nanophotonic motion detection system (67023). This accelerometer had measured sensitivity of approximately 10 nano-g. The Integrated NEMS and optoelectronics LDRD supported the nano-g accelerometer LDRD by providing advanced designs for the accelerometers, packaging, and a detection scheme to encapsulate the accelerometer, furthering the testing capabilities beyond bench-top tests. A fully packaged and tested die was never realized, but significant packaging issues were addressed and many resolved. The second technology supported by this work was the ultrasensitive directional microphone arrays for military operations in urban terrain and future combat systems (93518). This application utilized a diffraction-based sensing technique with different optical component placement and a different detection scheme from the nano-g accelerometer. The Integrated NEMS LDRD supported the microphone array LDRD by providing custom designs, VCSELs, and measurement techniques to accelerometers that were fabricated from the same operational principles as the microphones, but contain proof masses for acceleration transduction. These devices were packaged at the end of the work.

Layout Optimisation of Wave Energy Converter Arrays

DEFF Research Database (Denmark)

Ruiz, Pau Mercadé; Nava, Vincenzo; Topper, Mathew B. R.

2017-01-01

This paper proposes an optimisation strategy for the layout design of wave energy converter (WEC) arrays. Optimal layouts are sought so as to maximise the absorbed power given a minimum q-factor, the minimum distance between WECs, and an area of deployment. To guarantee an efficient optimisation......, a four-parameter layout description is proposed. Three different optimisation algorithms are further compared in terms of performance and computational cost. These are the covariance matrix adaptation evolution strategy (CMA), a genetic algorithm (GA) and the glowworm swarm optimisation (GSO) algorithm...

Wake Vortex Avoidance System and Method

Science.gov (United States)

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

2017-01-01

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

A total generalized variation approach for near-field acoustic holography

DEFF Research Database (Denmark)

Fernandez Grande, Efren

2017-01-01

Near-field methods based on microphone array measurements are useful to understand how a source radiates sound. Due to discretization errors, these methods are typically restricted to low frequencies. Sparse approaches have gained considerable attention, as they can potentially recover a seemingl...

Three-dimensional lithographically-defined organotypic tissue arrays for quantitative analysis of morphogenesis and neoplastic progression

Energy Technology Data Exchange (ETDEWEB)

Nelson, Celeste M.; Inman, Jamie L.; Bissell, Mina J.

2008-02-13

Here we describe a simple micromolding method to construct three-dimensional arrays of organotypic epithelial tissue structures that approximate in vivo histology. An elastomeric stamp containing an array of posts of defined geometry and spacing is used to mold microscale cavities into the surface of type I collagen gels. Epithelial cells are seeded into the cavities and covered with a second layer of collagen. The cells reorganize into hollow tissues corresponding to the geometry of the cavities. Patterned tissue arrays can be produced in 3-4 h and will undergo morphogenesis over the following one to three days. The protocol can easily be adapted to study a variety of tissues and aspects of normal and neoplastic development.

Adaptive Jacobian Fuzzy Attitude Control for Flexible Spacecraft Combined Attitude and Sun Tracking System

Science.gov (United States)

Chak, Yew-Chung; Varatharajoo, Renuganth

2016-07-01

Many spacecraft attitude control systems today use reaction wheels to deliver precise torques to achieve three-axis attitude stabilization. However, irrecoverable mechanical failure of reaction wheels could potentially lead to mission interruption or total loss. The electrically-powered Solar Array Drive Assemblies (SADA) are usually installed in the pitch axis which rotate the solar arrays to track the Sun, can produce torques to compensate for the pitch-axis wheel failure. In addition, the attitude control of a flexible spacecraft poses a difficult problem. These difficulties include the strong nonlinear coupled dynamics between the rigid hub and flexible solar arrays, and the imprecisely known system parameters, such as inertia matrix, damping ratios, and flexible mode frequencies. In order to overcome these drawbacks, the adaptive Jacobian tracking fuzzy control is proposed for the combined attitude and sun-tracking control problem of a flexible spacecraft during attitude maneuvers in this work. For the adaptation of kinematic and dynamic uncertainties, the proposed scheme uses an adaptive sliding vector based on estimated attitude velocity via approximate Jacobian matrix. The unknown nonlinearities are approximated by deriving the fuzzy models with a set of linguistic If-Then rules using the idea of sector nonlinearity and local approximation in fuzzy partition spaces. The uncertain parameters of the estimated nonlinearities and the Jacobian matrix are being adjusted online by an adaptive law to realize feedback control. The attitude of the spacecraft can be directly controlled with the Jacobian feedback control when the attitude pointing trajectory is designed with respect to the spacecraft coordinate frame itself. A significant feature of this work is that the proposed adaptive Jacobian tracking scheme will result in not only the convergence of angular position and angular velocity tracking errors, but also the convergence of estimated angular velocity to

Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation

Directory of Open Access Journals (Sweden)

Schüldt Christian

2007-01-01

Full Text Available An acoustic echo cancellation structure with a single loudspeaker and multiple microphones is, from a system identification perspective, generally modelled as a single-input multiple-output system. Such a system thus implies specific echo-path models (adaptive filter for every loudspeaker to microphone path. Due to the often large dimensionality of the filters, which is required to model rooms with standard reverberation time, the adaptation process can be computationally demanding. This paper presents a selective updating normalized least mean square (NLMS-based method which reduces complexity to nearly half in practical situations, while showing superior convergence speed performance as compared to conventional complexity reduction schemes. Moreover, the method concentrates the filter adaptation to the filter which is most misadjusted, which is a typically desired feature.

Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation

Directory of Open Access Journals (Sweden)

Fredric Lindstrom

2007-02-01

Full Text Available An acoustic echo cancellation structure with a single loudspeaker and multiple microphones is, from a system identification perspective, generally modelled as a single-input multiple-output system. Such a system thus implies specific echo-path models (adaptive filter for every loudspeaker to microphone path. Due to the often large dimensionality of the filters, which is required to model rooms with standard reverberation time, the adaptation process can be computationally demanding. This paper presents a selective updating normalized least mean square (NLMS-based method which reduces complexity to nearly half in practical situations, while showing superior convergence speed performance as compared to conventional complexity reduction schemes. Moreover, the method concentrates the filter adaptation to the filter which is most misadjusted, which is a typically desired feature.

Phased-array technology for automatic pipeline inspection; Phased Array-Technologie fuer automatisierte Pipeline-Inspektion

Energy Technology Data Exchange (ETDEWEB)

Bosch, J.; Hugger, A.; Franz, J. [GE Energy, PII Pipetronix GmbH, Stutensee (Germany); Falter, S.; Oberdoerfer, Y. [GE Inspection Technology Systems, Huerth (Germany)

2004-07-01

Pipeline inspection pigs with individual test probes are limited in their function due to the fixed arrangement of sensors on the support. In contrast, the phased-array technology enables multitasking of tests, e.g. stress and corrosion testing which formerly required two different test runs with different sensor set-ups. The angles of inclination can be adapted to the test medium, and virtual sensors can be matched in size and overlap so that, e.g., small pittings will be detected. The sensor set-up presented here enables higher test speed and improved flaw detection. The contribution describes the measuring principle, the inspection pig (UltraScan DUO), and some results of prototype measurements. [German] Pruefmolche fuer die Pipelinepruefung mit Einzelpruefkoepfen sind in ihrem Funktionsumfang aufgrund der festliegenden Anordnung der Sensoren im Sensortraeger eingeschraenkt. Die Phased-Array-Technologie gestattet die simultane Durchfuehrung verschiedener Pruefaufgaben, wie beispielsweise der Rissund der Korrosionspruefung, die vorher zwei Prueflaeufe mit verschiedenen Sensortraegern erforderten. Die Einfallswinkel koennen auf das jeweilige Medium angepasst werden, und es besteht die Moeglichkeit, virtuelle Sensoren bezueglich ihrer Groesse und der gegenseitigen Ueberlappung so anzupassen, dass beispielsweise kleine Pittings gefunden werden koennen. Die ausgefuehrte Form gestattet hoehere Pruefgeschwindigkeit und verbesserte Fehlerauffindung. In diesem Artikel werden das Messprinzip und der Inspektionsmolch (UltraScan DUO) beschrieben sowie einige Prototyp-Messergebnisse vorgestellt.

Riems influenza a typing array (RITA): An RT-qPCR-based low density array for subtyping avian and mammalian influenza a viruses.

Science.gov (United States)

Hoffmann, Bernd; Hoffmann, Donata; Henritzi, Dinah; Beer, Martin; Harder, Timm C

2016-06-03

Rapid and sensitive diagnostic approaches are of the utmost importance for the detection of humans and animals infected by specific influenza virus subtype(s). Cascade-like diagnostics starting with the use of pan-influenza assays and subsequent subtyping devices are normally used. Here, we demonstrated a novel low density array combining 32 TaqMan(®) real-time RT-PCR systems in parallel for the specific detection of the haemagglutinin (HA) and neuraminidase (NA) subtypes of avian and porcine hosts. The sensitivity of the newly developed system was compared with that of the pan-influenza assay, and the specificity of all RT-qPCRs was examined using a broad panel of 404 different influenza A virus isolates representing 45 different subtypes. Furthermore, we analysed the performance of the RT-qPCR assays with diagnostic samples obtained from wild birds and swine. Due to the open format of the array, adaptations to detect newly emerging influenza A virus strains can easily be integrated. The RITA array represents a competitive, fast and sensitive subtyping tool that requires neither new machinery nor additional training of staff in a lab where RT-qPCR is already established.

The sound of high winds. The effect of atmospheric stability on wind turbine sound and microphone noise

International Nuclear Information System (INIS)

Van den Berg, G.P.

2006-01-01

In this thesis issues are raised concerning wind turbine noise and its relationship to altitude dependent wind velocity. The following issues are investigated: what is the influence of atmospheric stability on the speed and sound power of a wind turbine?; what is the influence of atmospheric stability on the character of wind turbine sound?; how widespread is the impact of atmospheric stability on wind turbine performance: is it relevant for new wind turbine projects; how can noise prediction take this stability into account?; what can be done to deal with the resultant higher impact of wind turbine sound? Apart from these directly wind turbine related issues, a final aim was to address a measurement problem: how does wind on a microphone affect the measurement of the ambient sound level?

Simulation tools for industrial applications of phased array inspection techniques

International Nuclear Information System (INIS)

Mahaut, St.; Roy, O.; Chatillon, S.; Calmon, P.

2001-01-01

Ultrasonic phased arrays techniques have been developed at the French Atomic Energy Commission in order to improve defects characterization and adaptability to various inspection configuration (complex geometry specimen). Such transducers allow 'standard' techniques - adjustable beam-steering and focusing -, or more 'advanced' techniques - self-focusing on defects for instance -. To estimate the performances of those techniques, models have been developed, which allows to compute the ultrasonic field radiated by an arbitrary phased array transducer through any complex specimen, and to predict the ultrasonic response of various defects inspected with a known beam. Both modeling applications are gathered in the Civa software, dedicated to NDT expertise. The use of those complementary models allows to evaluate the ability of a phased array to steer and focus the ultrasonic beam, and therefore its relevancy to detect and characterize defects. These models are specifically developed to give accurate solutions to realistic inspection applications. This paper briefly describes the CIVA models, and presents some applications dedicated to the inspection of complex specimen containing various defects with a phased array used to steer and focus the beam. Defect detection and characterization performances are discussed for the various configurations. Some experimental validation of both models are also presented. (authors)

Unvoiced Speech Recognition Using Tissue-Conductive Acoustic Sensor

Directory of Open Access Journals (Sweden)

Hiroshi Saruwatari

2007-01-01

Full Text Available We present the use of stethoscope and silicon NAM (nonaudible murmur microphones in automatic speech recognition. NAM microphones are special acoustic sensors, which are attached behind the talker's ear and can capture not only normal (audible speech, but also very quietly uttered speech (nonaudible murmur. As a result, NAM microphones can be applied in automatic speech recognition systems when privacy is desired in human-machine communication. Moreover, NAM microphones show robustness against noise and they might be used in special systems (speech recognition, speech transform, etc. for sound-impaired people. Using adaptation techniques and a small amount of training data, we achieved for a 20 k dictation task a 93.9% word accuracy for nonaudible murmur recognition in a clean environment. In this paper, we also investigate nonaudible murmur recognition in noisy environments and the effect of the Lombard reflex on nonaudible murmur recognition. We also propose three methods to integrate audible speech and nonaudible murmur recognition using a stethoscope NAM microphone with very promising results.

SNP Arrays

Directory of Open Access Journals (Sweden)

Jari Louhelainen

2016-10-01

Full Text Available 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.

A novel method to design sparse linear arrays for ultrasonic phased array.

Science.gov (United States)

Yang, Ping; Chen, Bin; Shi, Ke-Ren

2006-12-22

In ultrasonic phased array testing, a sparse array can increase the resolution by enlarging the aperture without adding system complexity. Designing a sparse array involves choosing the best or a better configuration from a large number of candidate arrays. We firstly designed sparse arrays by using a genetic algorithm, but found that the arrays have poor performance and poor consistency. So, a method based on the Minimum Redundancy Linear Array was then adopted. Some elements are determined by the minimum-redundancy array firstly in order to ensure spatial resolution and then a genetic algorithm is used to optimize the remaining elements. Sparse arrays designed by this method have much better performance and consistency compared to the arrays designed only by a genetic algorithm. Both simulation and experiment confirm the effectiveness.

Robotic inspection of fiber reinforced composites using phased array UT

Science.gov (United States)

Stetson, Jeffrey T.; De Odorico, Walter

2014-02-01

Ultrasound is the current NDE method of choice to inspect large fiber reinforced airframe structures. Over the last 15 years Cartesian based scanning machines using conventional ultrasound techniques have been employed by all airframe OEMs and their top tier suppliers to perform these inspections. Technical advances in both computing power and commercially available, multi-axis robots now facilitate a new generation of scanning machines. These machines use multiple end effector tools taking full advantage of phased array ultrasound technologies yielding substantial improvements in inspection quality and productivity. This paper outlines the general architecture for these new robotic scanning systems as well as details the variety of ultrasonic techniques available for use with them including advances such as wide area phased array scanning and sound field adaptation for non-flat, non-parallel surfaces.

Free-floating epithelial micro-tissue arrays: a low cost and versatile technique.

Science.gov (United States)

Flood, P; Alvarez, L; Reynaud, E G

2016-10-11

Three-dimensional (3D) tissue models are invaluable tools that can closely reflect the in vivo physiological environment. However, they are usually difficult to develop, have a low throughput and are often costly; limiting their utility to most laboratories. The recent availability of inexpensive additive manufacturing printers and open source 3D design software offers us the possibility to easily create affordable 3D cell culture platforms. To demonstrate this, we established a simple, inexpensive and robust method for producing arrays of free-floating epithelial micro-tissues. Using a combination of 3D computer aided design and 3D printing, hydrogel micro-moulding and collagen cell encapsulation we engineered microenvironments that consistently direct the growth of micro-tissue arrays. We described the adaptability of this technique by testing several immortalised epithelial cell lines (MDCK, A549, Caco-2) and by generating branching morphology and micron to millimetre scaled micro-tissues. We established by fluorescence and electron microscopy that micro-tissues are polarised, have cell type specific differentiated phenotypes and regain native in vivo tissue qualities. Finally, using Salmonella typhimurium we show micro-tissues display a more physiologically relevant infection response compared to epithelial monolayers grown on permeable filter supports. In summary, we have developed a robust and adaptable technique for producing arrays of epithelial micro-tissues. This in vitro model has the potential to be a valuable tool for studying epithelial cell and tissue function/architecture in a physiologically relevant context.

A low-density SNP array for analyzing differential selection in freshwater and marine populations of threespine stickleback (Gasterosteus aculeatus).

Science.gov (United States)

Ferchaud, Anne-Laure; Pedersen, Susanne H; Bekkevold, Dorte; Jian, Jianbo; Niu, Yongchao; Hansen, Michael M

2014-10-06

The threespine stickleback (Gasterosteus aculeatus) has become an important model species for studying both contemporary and parallel evolution. In particular, differential adaptation to freshwater and marine environments has led to high differentiation between freshwater and marine stickleback populations at the phenotypic trait of lateral plate morphology and the underlying candidate gene Ectodysplacin (EDA). Many studies have focused on this trait and candidate gene, although other genes involved in marine-freshwater adaptation may be equally important. In order to develop a resource for rapid and cost efficient analysis of genetic divergence between freshwater and marine sticklebacks, we generated a low-density SNP (Single Nucleotide Polymorphism) array encompassing markers of chromosome regions under putative directional selection, along with neutral markers for background. RAD (Restriction site Associated DNA) sequencing of sixty individuals representing two freshwater and one marine population led to the identification of 33,993 SNP markers. Ninety-six of these were chosen for the low-density SNP array, among which 70 represented SNPs under putatively directional selection in freshwater vs. marine environments, whereas 26 SNPs were assumed to be neutral. Annotation of these regions revealed several genes that are candidates for affecting stickleback phenotypic variation, some of which have been observed in previous studies whereas others are new. We have developed a cost-efficient low-density SNP array that allows for rapid screening of polymorphisms in threespine stickleback. The array provides a valuable tool for analyzing adaptive divergence between freshwater and marine stickleback populations beyond the well-established candidate gene Ectodysplacin (EDA).

Improvements on Fresnel arrays for high contrast imaging

Science.gov (United States)

Wilhem, Roux; Laurent, Koechlin

2018-03-01

The Fresnel Diffractive Array Imager (FDAI) is based on a new optical concept for space telescopes, developed at Institut de Recherche en Astrophysique et Planétologie (IRAP), Toulouse, France. For the visible and near-infrared it has already proven its performances in resolution and dynamic range. We propose it now for astrophysical applications in the ultraviolet with apertures from 6 to 30 meters, aimed at imaging in UV faint astrophysical sources close to bright ones, as well as other applications requiring high dynamic range. Of course the project needs first a probatory mission at small aperture to validate the concept in space. In collaboration with institutes in Spain and Russia, we will propose to board a small prototype of Fresnel imager on the International Space Station (ISS), with a program combining technical tests and astrophysical targets. The spectral domain should contain the Lyman- α line ( λ = 121 nm). As part of its preparation, we improve the Fresnel array design for a better Point Spread Function in UV, presently on a small laboratory prototype working at 260 nm. Moreover, we plan to validate a new optical design and chromatic correction adapted to UV. In this article we present the results of numerical propagations showing the improvement in dynamic range obtained by combining and adapting three methods : central obturation, optimization of the bars mesh holding the Fresnel rings, and orthogonal apodization. We briefly present the proposed astrophysical program of a probatory mission with such UV optics.

Ultrasonic phased arrays for nondestructive inspection of forgings

International Nuclear Information System (INIS)

Wuestenberg, H.; Rotter, B.; Klanke, H.P.; Harbecke, D.

1993-01-01

Ultrasonic examinations on large forgings like rotor shafts for turbines or components for nuclear reactors are carried out at various manufacturing stages and during in-service inspections. During the manufacture, most of the inspections are carried out manually. Special in-service conditions, such as those at nuclear pressure vessels, have resulted in the development of mechanized scanning equipment. Ultrasonic probes have improved, and well-adapted sound fields and pulse shapes and based on special imaging procedures for the representation of the reportable reflectors have been applied. Since the geometry of many forgings requires the use of a multitude of angles for the inspections in-service and during manufacture, phased-array probes can be used successfully. The main advantages of the phased-array concept, e.g. the generation of a multitude of angles with the typical increase of redundancy in detection and quantitative evaluation and the possibility to produce pictures of defect situations, will be described in this contribution

Multiengine Speech Processing Using SNR Estimator in Variable Noisy Environments

Directory of Open Access Journals (Sweden)

Ahmad R. Abu-El-Quran

2012-01-01

Full Text Available We introduce a multiengine speech processing system that can detect the location and the type of audio signal in variable noisy environments. This system detects the location of the audio source using a microphone array; the system examines the audio first, determines if it is speech/nonspeech, then estimates the value of the signal to noise (SNR using a Discrete-Valued SNR Estimator. Using this SNR value, instead of trying to adapt the speech signal to the speech processing system, we adapt the speech processing system to the surrounding environment of the captured speech signal. In this paper, we introduced the Discrete-Valued SNR Estimator and a multiengine classifier, using Multiengine Selection or Multiengine Weighted Fusion. Also we use the SI as example of the speech processing. The Discrete-Valued SNR Estimator achieves an accuracy of 98.4% in characterizing the environment's SNR. Compared to a conventional single engine SI system, the improvement in accuracy was as high as 9.0% and 10.0% for the Multiengine Selection and Multiengine Weighted Fusion, respectively.

Microbial Warfare: Illuminating CRISPR adaptive immunity using single-molecule fluorescence

NARCIS (Netherlands)

Loeff, L.

2017-01-01

Bacteria and archaea are constantly threatened by a large array of viruses and other genetic elements. Driven by evolution, these organisms have acquired a wide arsenal of defense mechanisms that allow the host organism to fight off the invaders. Among these defense mechanisms is an adaptive and

Wideband Low Side Lobe Aperture Coupled Patch Phased Array Antennas

Science.gov (United States)

Poduval, Dhruva

Low profile printed antenna arrays with wide bandwidth, high gain, and low Side Lobe Level (SLL) are in great demand for current and future commercial and military communication systems and radar. Aperture coupled patch antennas have been proposed to obtain wide impedance bandwidths in the past. Aperture coupling is preferred particularly for phased arrays because of their advantage of integration to other active devices and circuits, e.g. phase shifters, power amplifiers, low noise amplifiers, mixers etc. However, when designing such arrays, the interplay between array performance characteristics, such as gain, side lobe level, back lobe level, mutual coupling etc. must be understood and optimized under multiple design constraints, e.g. substrate material properties and thicknesses, element to element spacing, and feed lines and their orientation and arrangements with respect to the antenna elements. The focus of this thesis is to investigate, design, and develop an aperture coupled patch array with wide operating bandwidth (30%), high gain (17.5 dBi), low side lobe level (20 dB), and high Forward to Backward (F/B) ratio (21.8 dB). The target frequency range is 2.4 to 3 GHz given its wide application in WLAN, LTE (Long Term Evolution) and other communication systems. Notwithstanding that the design concept can very well be adapted at other frequencies. Specifically, a 16 element, 4 by 4 planar microstrip patch array is designed using HFSS and experimentally developed and tested. Starting from mutual coupling minimization a corporate feeding scheme is designed to achieve the needed performance. To reduce the SLL the corporate feeding network is redesigned to obtain a specific amplitude taper. Studies are conducted to determine the optimum location for a metallic reflector under the feed line to improve the F/B. An experimental prototype of the antenna was built and tested validating and demonstrating the performance levels expected from simulation predictions

Analysis and Design of a Context Adaptable SAD/MSE Architecture

Directory of Open Access Journals (Sweden)

Arvind Sudarsanam

2009-01-01

Full Text Available Design of flexible multimedia accelerators that can cater to multiple algorithms is being aggressively pursued in the media processors community. Such an approach is justified in the era of sub-45 nm technology where an increasingly dominating leakage power component is forcing designers to make the best possible use of on-chip resources. In this paper we present an analysis of two commonly used window-based operations (sum of absolute differences and mean squared error across a variety of search patterns and block sizes (2×3, 5×5, etc.. We propose a context adaptable architecture that has (i configurable 2D systolic array and (ii 2D Configurable Register Array (CRA. CRA can cater to variable pixel access patterns while reusing fetched pixels across search windows. Benefits of proposed architecture when compared to 15 other published architectures are adaptability, high throughput, and low latency at a cost of increased footprint, when ported on a Xilinx FPGA.

Monitor, a Vibrotactile Aid for Environmental Perception: A Field Evaluation by Four People with Severe Hearing and Vision Impairment

Directory of Open Access Journals (Sweden)

Parivash Ranjbar

2013-01-01

Full Text Available Monitor is a portable vibrotactile aid to improve the ability of people with severe hearing impairment or deafblindness to detect, identify, and recognize the direction of sound-producing events. It transforms and adapts sounds to the frequency sensitivity range of the skin. The aid was evaluated in the field. Four females (44–54 years with Usher Syndrome I (three with tunnel vision and one with only light perception tested the aid at home and in traffic in three different field studies: without Monitor, with Monitor with an omnidirectional microphone, and with Monitor with a directional microphone. The tests were video-documented, and the two field studies with Monitor were initiated after five weeks of training. The detection scores with omnidirectional and directional microphones were 100% for three participants and above 57% for one, both in their home and traffic environments. In the home environment the identification scores with the omnidirectional microphone were 70%–97% and 58%–95% with the directional microphone. The corresponding values in traffic were 29%–100% and 65%–100%, respectively. Their direction perception was improved to some extent by both microphones. Monitor improved the ability of people with deafblindness to detect, identify, and recognize the direction of events producing sounds.

Monitor, a vibrotactile aid for environmental perception: a field evaluation by four people with severe hearing and vision impairment.

Science.gov (United States)

Ranjbar, Parivash; Stenström, Ingeborg

2013-01-01

Monitor is a portable vibrotactile aid to improve the ability of people with severe hearing impairment or deafblindness to detect, identify, and recognize the direction of sound-producing events. It transforms and adapts sounds to the frequency sensitivity range of the skin. The aid was evaluated in the field. Four females (44-54 years) with Usher Syndrome I (three with tunnel vision and one with only light perception) tested the aid at home and in traffic in three different field studies: without Monitor, with Monitor with an omnidirectional microphone, and with Monitor with a directional microphone. The tests were video-documented, and the two field studies with Monitor were initiated after five weeks of training. The detection scores with omnidirectional and directional microphones were 100% for three participants and above 57% for one, both in their home and traffic environments. In the home environment the identification scores with the omnidirectional microphone were 70%-97% and 58%-95% with the directional microphone. The corresponding values in traffic were 29%-100% and 65%-100%, respectively. Their direction perception was improved to some extent by both microphones. Monitor improved the ability of people with deafblindness to detect, identify, and recognize the direction of events producing sounds.

Simulation of an Electromagnetic Acoustic Transducer Array by Using Analytical Method and FDTD

Directory of Open Access Journals (Sweden)

Yuedong Xie

2016-01-01

Full Text Available Previously, we developed a method based on FEM and FDTD for the study of an Electromagnetic Acoustic Transducer Array (EMAT. This paper presents a new analytical solution to the eddy current problem for the meander coil used in an EMAT, which is adapted from the classic Deeds and Dodd solution originally intended for circular coils. The analytical solution resulting from this novel adaptation exploits the large radius extrapolation and shows several advantages over the finite element method (FEM, especially in the higher frequency regime. The calculated Lorentz force density from the analytical EM solver is then coupled to the ultrasonic simulations, which exploit the finite-difference time-domain (FDTD method to describe the propagation of ultrasound waves, in particular for Rayleigh waves. Radiation pattern obtained with Hilbert transform on time-domain waveforms is proposed to characterise the sensor in terms of its beam directivity and field distribution along the steering angle, which can produce performance parameters for an EMAT array, facilitating the optimum design of such sensors.

Millimetre Level Accuracy GNSS Positioning with the Blind Adaptive Beamforming Method in Interference Environments

Directory of Open Access Journals (Sweden)

Saeed Daneshmand

2016-10-01

Full Text Available The use of antenna arrays in Global Navigation Satellite System (GNSS applications is gaining significant attention due to its superior capability to suppress both narrowband and wideband interference. However, the phase distortions resulting from array processing may limit the applicability of these methods for high precision applications using carrier phase based positioning techniques. This paper studies the phase distortions occurring with the adaptive blind beamforming method in which satellite angle of arrival (AoA information is not employed in the optimization problem. To cater to non-stationary interference scenarios, the array weights of the adaptive beamformer are continuously updated. The effects of these continuous updates on the tracking parameters of a GNSS receiver are analyzed. The second part of this paper focuses on reducing the phase distortions during the blind beamforming process in order to allow the receiver to perform carrier phase based positioning by applying a constraint on the structure of the array configuration and by compensating the array uncertainties. Limitations of the previous methods are studied and a new method is proposed that keeps the simplicity of the blind beamformer structure and, at the same time, reduces tracking degradations while achieving millimetre level positioning accuracy in interference environments. To verify the applicability of the proposed method and analyze the degradations, array signals corresponding to the GPS L1 band are generated using a combination of hardware and software simulators. Furthermore, the amount of degradation and performance of the proposed method under different conditions are evaluated based on Monte Carlo simulations.

The Contribution of Genetic Recombination to CRISPR Array Evolution.

Science.gov (United States)

Kupczok, Anne; Landan, Giddy; Dagan, Tal

2015-06-16

CRISPR (clustered regularly interspaced short palindromic repeats) is a microbial immune system against foreign DNA. Recognition sequences (spacers) encoded within the CRISPR array mediate the immune reaction in a sequence-specific manner. The known mechanisms for the evolution of CRISPR arrays include spacer acquisition from foreign DNA elements at the time of invasion and array erosion through spacer deletion. Here, we consider the contribution of genetic recombination between homologous CRISPR arrays to the evolution of spacer repertoire. Acquisition of spacers from exogenic arrays via recombination may confer the recipient with immunity against unencountered antagonists. For this purpose, we develop a novel method for the detection of recombination in CRISPR arrays by modeling the spacer order in arrays from multiple strains from the same species. Because the evolutionary signal of spacer recombination may be similar to that of pervasive spacer deletions or independent spacer acquisition, our method entails a robustness analysis of the recombination inference by a statistical comparison to resampled and perturbed data sets. We analyze CRISPR data sets from four bacterial species: two Gammaproteobacteria species harboring CRISPR type I and two Streptococcus species harboring CRISPR type II loci. We find that CRISPR array evolution in Escherichia coli and Streptococcus agalactiae can be explained solely by vertical inheritance and differential spacer deletion. In Pseudomonas aeruginosa, we find an excess of single spacers potentially incorporated into the CRISPR locus during independent acquisition events. In Streptococcus thermophilus, evidence for spacer acquisition by recombination is present in 5 out of 70 strains. Genetic recombination has been proposed to accelerate adaptation by combining beneficial mutations that arose in independent lineages. However, for most species under study, we find that CRISPR evolution is shaped mainly by spacer acquisition and

Aeroacoustic measurements for an axial fan in a non-anechoic environment

International Nuclear Information System (INIS)

Davoudi, Behdad; Foss, John F; Morris, Scott C

2016-01-01

Determination of the aeroacoustic emission from an axial fan in a non-anechoic environment is a challenging experimental task given ambient noise and acoustic reflections from surrounding objects. Successful strategies to address this task for a representative nine and three blade fan are presented. An array consisting of ten microphones was constructed and placed in the upstream region of the axial fans to measure the fan acoustic signature at ten distinct locations. A novel delay and sum (DS) beamforming technique (that allows precise time delays to be established by the use of cross correlation techniques) was applied to the microphone outputs in order to separate the fans’ acoustic emissions from the ambient noise and reflections from the facility walls. A numerical simulation was developed to represent the experimental facility and the measurements. The numerical simulation indicated that the extraneous noise can be satisfactorily separated from the fan noise using the array measurements and post processing the acoustic data with the present DS beamforming technique. (paper)

HIGH-SPEED IMAGING AND WAVEFRONT SENSING WITH AN INFRARED AVALANCHE PHOTODIODE ARRAY

Energy Technology Data Exchange (ETDEWEB)

Baranec, Christoph; Atkinson, Dani; Hall, Donald; Jacobson, Shane; Chun, Mark [Institute for Astronomy, University of Hawai‘i at Mānoa, Hilo, HI 96720-2700 (United States); Riddle, Reed [Division of Physics, Mathematics, and Astronomy, California Institute of Technology, Pasadena, CA 91125 (United States); Law, Nicholas M., E-mail: baranec@hawaii.edu [Department of Physics and Astronomy, University of North Carolina at Chapel Hill, Chapel Hill, NC 27599-3255 (United States)

2015-08-10

Infrared avalanche photodiode (APD) arrays represent a panacea for many branches of astronomy by enabling extremely low-noise, high-speed, and even photon-counting measurements at near-infrared wavelengths. We recently demonstrated the use of an early engineering-grade infrared APD array that achieves a correlated double sampling read noise of 0.73 e{sup −} in the lab, and a total noise of 2.52 e{sup −} on sky, and supports simultaneous high-speed imaging and tip-tilt wavefront sensing with the Robo-AO visible-light laser adaptive optics (AO) system at the Palomar Observatory 1.5 m telescope. Here we report on the improved image quality simultaneously achieved at visible and infrared wavelengths by using the array as part of an image stabilization control loop with AO-sharpened guide stars. We also discuss a newly enabled survey of nearby late M-dwarf multiplicity, as well as future uses of this technology in other AO and high-contrast imaging applications.

Maneuver Acoustic Flight Test of the Bell 430 Helicopter Data Report

Science.gov (United States)

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

2014-01-01

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

Coupling in reflector arrays

DEFF Research Database (Denmark)

Appel-Hansen, Jørgen

1968-01-01

In order to reduce the space occupied by a reflector array, it is desirable to arrange the array antennas as close to each other as possible; however, in this case coupling between the array antennas will reduce the reflecting properties of the reflector array. The purpose of the present communic......In order to reduce the space occupied by a reflector array, it is desirable to arrange the array antennas as close to each other as possible; however, in this case coupling between the array antennas will reduce the reflecting properties of the reflector array. The purpose of the present...

Comparisons of receive array interference reduction techniques under erroneous generalized transmit beamforming

KAUST Repository

Radaydeh, Redha Mahmoud

2014-02-01

This paper studies generalized single-stream transmit beamforming employing receive array co-channel interference reduction algorithms under slow and flat fading multiuser wireless systems. The impact of imperfect prediction of channel state information for the desired user spatially uncorrelated transmit channels on the effectiveness of transmit beamforming for different interference reduction techniques is investigated. The case of over-loaded receive array with closely-spaced elements is considered, wherein it can be configured to specified interfering sources. Both dominant interference reduction and adaptive interference reduction techniques for statistically ordered and unordered interferers powers, respectively, are thoroughly studied. The effect of outdated statistical ordering of the interferers powers on the efficiency of dominant interference reduction is studied and then compared against the adaptive interference reduction. For the system models described above, new analytical formulations for the statistics of combined signal-to-interference-plus-noise ratio are presented, from which results for conventional maximum ratio transmission and single-antenna best transmit selection can be directly deduced as limiting cases. These results are then utilized to obtain quantitative measures for various performance metrics. They are also used to compare the achieved performance of various configuration models under consideration. © 1972-2012 IEEE.

FPGA-Based Communications Receivers for Smart Antenna Array Embedded Systems

Directory of Open Access Journals (Sweden)

Millar James

2006-01-01

Full Text Available Field-programmable gate arrays (FPGAs are drawing ever increasing interest from designers of embedded wireless communications systems. They outpace digital signal processors (DSPs, through hardware execution of a wide range of parallelizable communications transceiver algorithms, at a fraction of the design and implementation effort and cost required for application-specific integrated circuits (ASICs. In our study, we employ an Altera Stratix FPGA development board, along with the DSP Builder software tool which acts as a high-level interface to the powerful Quartus II environment. We compare single- and multibranch FPGA-based receiver designs in terms of error rate performance and power consumption. We exploit FPGA operational flexibility and algorithm parallelism to design eigenmode-monitoring receivers that can adapt to variations in wireless channel statistics, for high-performing, inexpensive, smart antenna array embedded systems.

FPGA-Based Communications Receivers for Smart Antenna Array Embedded Systems

Directory of Open Access Journals (Sweden)

James Millar

2006-10-01

Full Text Available Field-programmable gate arrays (FPGAs are drawing ever increasing interest from designers of embedded wireless communications systems. They outpace digital signal processors (DSPs, through hardware execution of a wide range of parallelizable communications transceiver algorithms, at a fraction of the design and implementation effort and cost required for application-specific integrated circuits (ASICs. In our study, we employ an Altera Stratix FPGA development board, along with the DSP Builder software tool which acts as a high-level interface to the powerful Quartus II environment. We compare single- and multibranch FPGA-based receiver designs in terms of error rate performance and power consumption. We exploit FPGA operational flexibility and algorithm parallelism to design eigenmode-monitoring receivers that can adapt to variations in wireless channel statistics, for high-performing, inexpensive, smart antenna array embedded systems.

Genome architecture enables local adaptation of Atlantic cod despite high connectivity

DEFF Research Database (Denmark)

Barth, Julia M I; Berg, Paul R; Jonsson, Per R.

2017-01-01

Adaptation to local conditions is a fundamental process in evolution; however, mechanisms maintaining local adaptation despite high gene flow are still poorly understood. Marine ecosystems provide a wide array of diverse habitats that frequently promote ecological adaptation even in species...... characterized by strong levels of gene flow. As one example, populations of the marine fish Atlantic cod (Gadus morhua) are highly connected due to immense dispersal capabilities but nevertheless show local adaptation in several key traits. By combining population genomic analyses based on 12K single......-nucleotide polymorphisms with larval dispersal patterns inferred using a biophysical ocean model, we show that Atlantic cod individuals residing in sheltered estuarine habitats of Scandinavian fjords mainly belong to offshore oceanic populations with considerable connectivity between these diverse ecosystems. Nevertheless...

Conservation and adaptation to climate change.

Science.gov (United States)

Brooke, Cassandra

2008-12-01

The need to adapt to climate change has become increasingly apparent, and many believe the practice of biodiversity conservation will need to alter to face this challenge. Conservation organizations are eager to determine how they should adapt their practices to climate change. This involves asking the fundamental question of what adaptation to climate change means. Most studies on climate change and conservation, if they consider adaptation at all, assume it is equivalent to the ability of species to adapt naturally to climate change as stated in Article 2 of the United Nations Framework Convention on Climate Change. Adaptation, however, can refer to an array of activities that range from natural adaptation, at one end of the spectrum, to sustainability science in coupled human and natural systems at the other. Most conservation organizations deal with complex systems in which adaptation to climate change involves making decisions on priorities for biodiversity conservation in the face of dynamic risks and involving the public in these decisions. Discursive methods such as analytic deliberation are useful for integrating scientific knowledge with public perceptions and values, particularly when large uncertainties and risks are involved. The use of scenarios in conservation planning is a useful way to build shared understanding at the science-policy interface. Similarly, boundary organizations-organizations or institutions that bridge different scales or mediate the relationship between science and policy-could prove useful for managing the transdisciplinary nature of adaptation to climate change, providing communication and brokerage services and helping to build adaptive capacity. The fact that some nongovernmental organizations (NGOs) are active across the areas of science, policy, and practice makes them well placed to fulfill this role in integrated assessments of biodiversity conservation and adaptation to climate change.

An application of neural network for Structural Health Monitoring of an adaptive wing with an array of FBG sensors

International Nuclear Information System (INIS)

Mieloszyk, Magdalena; Skarbek, Lukasz; Ostachowicz, Wieslaw; Krawczuk, Marek

2011-01-01

This paper presents an application of neural networks to determinate the level of activation of shape memory alloy actuators of an adaptive wing. In this concept the shape of the wing can be controlled and altered thanks to the wing design and the use of integrated shape memory alloy actuators. The wing is assumed as assembled from a number of wing sections that relative positions can be controlled independently by thermal activation of shape memory actuators. The investigated wing is employed with an array of Fibre Bragg Grating sensors. The Fibre Bragg Grating sensors with combination of a neural network have been used to Structural Health Monitoring of the wing condition. The FBG sensors are a great tool to control the condition of composite structures due to their immunity to electromagnetic fields as well as their small size and weight. They can be mounted onto the surface or embedded into the wing composite material without any significant influence on the wing strength. The paper concentrates on analysis of the determination of the twisting moment produced by an activated shape memory alloy actuator. This has been analysed both numerically using the finite element method by a commercial code ABAQUS (registered) and experimentally using Fibre Bragg Grating sensor measurements. The results of the analysis have been then used by a neural network to determine twisting moments produced by each shape memory alloy actuator.

Concurrent array-based queue

Science.gov (United States)

Heidelberger, Philip; Steinmacher-Burow, Burkhard

2015-01-06

According to one embodiment, a method for implementing an array-based queue in memory of a memory system that includes a controller includes configuring, in the memory, metadata of the array-based queue. The configuring comprises defining, in metadata, an array start location in the memory for the array-based queue, defining, in the metadata, an array size for the array-based queue, defining, in the metadata, a queue top for the array-based queue and defining, in the metadata, a queue bottom for the array-based queue. The method also includes the controller serving a request for an operation on the queue, the request providing the location in the memory of the metadata of the queue.

Past and future detector arrays for complete event reconstruction in heavy-ion reactions

Science.gov (United States)

Cardella, G.; Acosta, L.; Auditore, L.; Boiano, C.; Castoldi, A.; D'Andrea, M.; De Filippo, E.; Dell'Aquila, D.; De Luca, S.; Fichera, F.; Giudice, N.; Gnoffo, B.; Grimaldi, A.; Guazzoni, C.; Lanzalone, G.; Librizzi, F.; Lombardo, I.; Maiolino, C.; Maffesanti, S.; Martorana, N. S.; Norella, S.; Pagano, A.; Pagano, E. V.; Papa, M.; Parsani, T.; Passaro, G.; Pirrone, S.; Politi, G.; Previdi, F.; Quattrocchi, L.; Rizzo, F.; Russotto, P.; Saccà, G.; Salemi, G.; Sciliberto, D.; Trifirò, A.; Trimarchi, M.; Vigilante, M.

2017-11-01

Complex and more and more complete detector arrays have been developed in the last two decades, or are in advanced design stage, in different laboratories. Such arrays are necessary to fully characterize nuclear reactions induced by stable and exotic beams. The need for contemporary detection of charged particles, and/or γ -rays, and/or neutrons, has been stressed in many fields of nuclear structure and reaction dynamics, with particular attention to the improvement of both high angular and energy resolution. Some examples of detection systems adapted to various energy ranges is discussed. Emphasis is given to the possible update of relatively old 4π detectors with new electronics and new detection methods.

Past and future detector arrays for complete event reconstruction in heavy-ion reactions

International Nuclear Information System (INIS)

Cardella, G.; Acosta, L.; Auditore, L.

2016-01-01

Complex and more and more complete detector arrays have been developed in the last two decades, or are in advanced design stage, in different laboratories. Such arrays are necessary to fully characterize nuclear reactions induced by stable and exotic beams. The need for contemporary detection of charged particles, and/or γ-rays, and/or neutrons, has been stressed in many fields of nuclear structure and reaction dynamics, with particular attention to the improvement of both high angular and energy resolution. Some examples of detection systems adapted to various energy ranges is discussed. Emphasis is given to the possible update of relatively old 4π detectors with new electronics and new detection methods.

Acoustic Localization with Infrasonic Signals

Science.gov (United States)

Threatt, Arnesha; Elbing, Brian

2015-11-01

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

A beamforming system based on the acousto-optic effect

DEFF Research Database (Denmark)

Torras Rosell, Antoni; Barrera Figueroa, Salvador; Jacobsen, Finn

2012-01-01

Beamforming techniques are usually based on microphone arrays. The present work uses a beam of light as a sensor element, and describes a beamforming system that locates sound sources based on the acousto-optic effect, this is, the interaction between sound and light. The use of light as a sensin...

Visualizing Sound Directivity via Smartphone Sensors

Science.gov (United States)

Hawley, Scott H.; McClain, Robert E., Jr.

2018-01-01

When Yang-Hann Kim received the Rossing Prize in Acoustics Education at the 2015 meeting of the Acoustical Society of America, he stressed the importance of offering visual depictions of sound fields when teaching acoustics. Often visualization methods require specialized equipment such as microphone arrays or scanning apparatus. We present a…

Programmable cellular arrays. Faults testing and correcting in cellular arrays

International Nuclear Information System (INIS)

Cercel, L.

1978-03-01

A review of some recent researches about programmable cellular arrays in computing and digital processing of information systems is presented, and includes both combinational and sequential arrays, with full arbitrary behaviour, or which can realize better implementations of specialized blocks as: arithmetic units, counters, comparators, control systems, memory blocks, etc. Also, the paper presents applications of cellular arrays in microprogramming, in implementing of a specialized computer for matrix operations, in modeling of universal computing systems. The last section deals with problems of fault testing and correcting in cellular arrays. (author)

Online real-time reconstruction of adaptive TSENSE with commodity CPU / GPU hardware

DEFF Research Database (Denmark)

Roujol, Sebastien; de Senneville, Baudouin; Vahala, E.

2009-01-01

A real-time reconstruction for adaptive TSENSE is presented that is optimized for MR-guidance of interventional procedures. The proposed method allows high frame-rate imaging with low image latencies, even when large coil arrays are employed and can be implemented on affordable commodity hardware....

Viruses are a dominant driver of protein adaptation in mammals.

Science.gov (United States)

Enard, David; Cai, Le; Gwennap, Carina; Petrov, Dmitri A

2016-05-17

Viruses interact with hundreds to thousands of proteins in mammals, yet adaptation against viruses has only been studied in a few proteins specialized in antiviral defense. Whether adaptation to viruses typically involves only specialized antiviral proteins or affects a broad array of virus-interacting proteins is unknown. Here, we analyze adaptation in ~1300 virus-interacting proteins manually curated from a set of 9900 proteins conserved in all sequenced mammalian genomes. We show that viruses (i) use the more evolutionarily constrained proteins within the cellular functions they interact with and that (ii) despite this high constraint, virus-interacting proteins account for a high proportion of all protein adaptation in humans and other mammals. Adaptation is elevated in virus-interacting proteins across all functional categories, including both immune and non-immune functions. We conservatively estimate that viruses have driven close to 30% of all adaptive amino acid changes in the part of the human proteome conserved within mammals. Our results suggest that viruses are one of the most dominant drivers of evolutionary change across mammalian and human proteomes.

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

Science.gov (United States)

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

2016-12-01

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

A decade of adaptive governance scholarship: synthesis and future directions

Directory of Open Access Journals (Sweden)

Brian C. Chaffin

2014-09-01

Full Text Available Adaptive governance is an emergent form of environmental governance that is increasingly called upon by scholars and practitioners to coordinate resource management regimes in the face of the complexity and uncertainty associated with rapid environmental change. Although the term "adaptive governance" is not exclusively applied to the governance of social-ecological systems, related research represents a significant outgrowth of literature on resilience, social-ecological systems, and environmental governance. We present a chronology of major scholarship on adaptive governance, synthesizing efforts to define the concept and identifying the array of governance concepts associated with transformation toward adaptive governance. Based on this synthesis, we define adaptive governance as a range of interactions between actors, networks, organizations, and institutions emerging in pursuit of a desired state for social-ecological systems. In addition, we identify and discuss ambiguities in adaptive governance scholarship such as the roles of adaptive management, crisis, and a desired state for governance of social-ecological systems. Finally, we outline a research agenda to examine whether an adaptive governance approach can become institutionalized under current legal frameworks and political contexts. We suggest a further investigation of the relationship between adaptive governance and the principles of good governance; the roles of power and politics in the emergence of adaptive governance; and potential interventions such as legal reform that may catalyze or enhance governance adaptations or transformation toward adaptive governance.

Comparison of printed glycan array, suspension array and ELISA in the detection of human anti-glycan antibodies.

Science.gov (United States)

Pochechueva, Tatiana; Jacob, Francis; Goldstein, Darlene R; Huflejt, Margaret E; Chinarev, Alexander; Caduff, Rosemarie; Fink, Daniel; Hacker, Neville; Bovin, Nicolai V; Heinzelmann-Schwarz, Viola

2011-12-01

Anti-glycan antibodies represent a vast and yet insufficiently investigated subpopulation of naturally occurring and adaptive antibodies in humans. Recently, a variety of glycan-based microarrays emerged, allowing high-throughput profiling of a large repertoire of antibodies. As there are no direct approaches for comparison and evaluation of multi-glycan assays we compared three glycan-based immunoassays, namely printed glycan array (PGA), fluorescent microsphere-based suspension array (SA) and ELISA for their efficacy and selectivity in profiling anti-glycan antibodies in a cohort of 48 patients with and without ovarian cancer. The ABO blood group glycan antigens were selected as well recognized ligands for sensitivity and specificity assessments. As another ligand we selected P(1), a member of the P blood group system recently identified by PGA as a potential ovarian cancer biomarker. All three glyco-immunoassays reflected the known ABO blood groups with high performance. In contrast, anti-P(1) antibody binding profiles displayed much lower concordance. Whilst anti-P(1) antibody levels between benign controls and ovarian cancer patients were significantly discriminated using PGA (p=0.004), we got only similar results using SA (p=0.03) but not for ELISA. Our findings demonstrate that whilst assays were largely positively correlated, each presents unique characteristic features and should be validated by an independent patient cohort rather than another array technique. The variety between methods presumably reflects the differences in glycan presentation and the antigen/antibody ratio, assay conditions and detection technique. This indicates that the glycan-antibody interaction of interest has to guide the assay selection. © The Author(s) 2011. This article is published with open access at Springerlink.com

The rational for a mid-scala electrode array.

Science.gov (United States)

Boyle, P J

2016-06-01

Today increasing numbers of cochlear implant candidates have residual hearing that can be aided and hence is worth trying to preserve. This means that surgical technique and electrode array design must be adapted to minimize trauma. Wide opening of the round window is often preferred to reduce drill related trauma and to avoid pressure spikes during electrode array insertion. A recent meta-analysis suggested that there is no significant correlation between hearing preservation and either insertion depth or scala position. However, a slow insertion speed of at least 30seconds was associated with better hearing preservation. An electrode design is proposed that targets the middle of the scala tympani. This minimizes frictional forces from either lateral or medial wall during insertion and imposes less static pressure on cochlear structures following insertion. The flexibility to insert via the round window requires a 0.7-mm maximum dimension at the proximal end of the array. Micro-anatomical analysis by micro-CT indicated that a 420-degree insertion depth was optimal between cochlear coverage and available space within the scala tympani. Physical measurements showed that mean insertion forces remained below 10mN during insertion. A series of 20 human temporal bone insertions found a mean insertion depth of 400 degrees with no scala dislocations. Six clinical series, in total 94 cases, found postoperative hearing in 81% of cases with a mean loss of 12dB compared to preoperative levels. Speech understanding out to one year post-fitting trended better for a mid-scala design group than for a straight electrode array group; although the differences were not statistically significant. A mid-scala array design appears able to be inserted with minimal trauma, to return a predictable insertion depth across various sizes of cochleae and to support reasonable levels of speech understanding without relying on residual hearing. Copyright © 2016. Published by Elsevier Masson

A two-microphone noise reduction system for cochlear implant users with nearby microphones. Part II: Performance Evaluation

OpenAIRE

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

2008-01-01

Users of cochlear implants (auditory aids, which stimulate the auditory nerve electrically at the inner ear) often suffer from poor speech understanding in noise. We evaluate a small (intermicrophone distance 7 mm) and computationally inexpensive adaptive noise reduction system suitable for behind-the-ear cochlear implant speech processors. The system is evaluated in simulated and real, anechoic and reverberant environments. Results from simulations show improvements of 3.4 to 9.3 dB in signa...

Demosaicking algorithm for the Kodak-RGBW color filter array

Science.gov (United States)

Rafinazari, M.; Dubois, E.

2015-01-01

Digital cameras capture images through different Color Filter Arrays and then reconstruct the full color image. Each CFA pixel only captures one primary color component; the other primary components will be estimated using information from neighboring pixels. During the demosaicking algorithm, the two unknown color components will be estimated at each pixel location. Most of the demosaicking algorithms use the RGB Bayer CFA pattern with Red, Green and Blue filters. The least-Squares Luma-Chroma demultiplexing method is a state of the art demosaicking method for the Bayer CFA. In this paper we develop a new demosaicking algorithm using the Kodak-RGBW CFA. This particular CFA reduces noise and improves the quality of the reconstructed images by adding white pixels. We have applied non-adaptive and adaptive demosaicking method using the Kodak-RGBW CFA on the standard Kodak image dataset and the results have been compared with previous work.

Intense echolocation calls from two ;whispering' bats, Artibeus jamaicensis and Macrophyllum macrophyllum (Phyllostomidae)

DEFF Research Database (Denmark)

Brinkløv, Signe; Kalko, Elisabeth K V; Surlykke, Annemarie

2009-01-01

Bats use echolocation to exploit a variety of habitats and food types. Much research has documented how frequency-time features of echolocation calls are adapted to acoustic constraints imposed by habitat and prey but emitted sound intensities have received little attention. Bats from the family...... of Phyllostomidae have been categorised as low intensity (whispering) gleaners, assumed to emit echolocation calls with low source levels (approximately 70 dB SPL measured 10 cm from the bat's mouth). We used a multi-microphone array to determine intensities emitted from two phyllostomid bats from Panamá...... room. Both species emitted surprisingly intense signals with maximum source levels of 105 dB SPL r.m.s. for M. macrophyllum and 110 dB SPL r.m.s. for A. jamaicensis, hence much louder than a ;whisper'. M. macrophyllum was consistently loud (mean source level 101 dB SPL) whereas A. jamaicensis showed...

Cyclotron-Resonance-Maser Arrays

International Nuclear Information System (INIS)

Kesar, A.; Lei, L.; Dikhtyar, V.; Korol, M.; Jerby, E.

1999-01-01

The cyclotron-resonance-maser (CRM) array [1] is a radiation source which consists of CRM elements coupled together under a common magnetic field. Each CRM-element employs a low-energy electron-beam which performs a cyclotron interaction with the local electromagnetic wave. These waves can be coupled together among the CRM elements, hence the interaction is coherently synchronized in the entire array. The implementation of the CRM-array approach may alleviate several technological difficulties which impede the development of single-beam gyro-devices. Furthermore, it proposes new features, such as the phased-array antenna incorporated in the CRM-array itself. The CRM-array studies may lead to the development of compact, high-power radiation sources operating at low-voltages. This paper introduces new conceptual schemes of CRM-arrays, and presents the progress in related theoretical and experimental studies in our laboratory. These include a multi-mode analysis of a CRM-array, and a first operation of this device with five carbon-fiber cathodes