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Sample records for linear noise-cancelling microphone

  1. 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

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

    Science.gov (United States)

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

    2011-01-01

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

  3. High Dynamic Range RF Front End with Noise Cancellation and Linearization for WiMAX Receivers

    Directory of Open Access Journals (Sweden)

    J.-M. Wu

    2012-06-01

    Full Text Available This research deals with verification of the high dynamic range for a heterodyne radio frequency (RF front end. A 2.6 GHz RF front end is designed and implemented in a hybrid microwave integrated circuit (HMIC for worldwide interoperability for microwave access (WiMAX receivers. The heterodyne RF front end consists of a low-noise amplifier (LNA with noise cancellation, an RF bandpass filter (BPF, a downconverter with linearization, and an intermediate frequency (IF BPF. A noise canceling technique used in the low-noise amplifier eliminates a thermal noise and then reduces the noise figure (NF of the RF front end by 0.9 dB. Use of a downconverter with diode linearizer also compensates for gain compression, which increases the input-referred third-order intercept point (IIP3 of the RF front end by 4.3 dB. The proposed method substantially increases the spurious-free dynamic range (DRf of the RF front end by 3.5 dB.

  4. An adaptive noise cancelling system used for beam control at the Stanford Linear Accelerator Center

    International Nuclear Information System (INIS)

    Himel, T.; Allison, S.; Grossberg, P.; Hendrickson, L.; Sass, R.; Shoaee, H.

    1993-06-01

    The SLAC Linear Collider now has a total of twenty-four beam-steering feedback loops used to keep the electron and positron beams on their desired trajectories. Seven of these loops measure and control the same beam as it proceeds down the linac through the arcs to the final focus. Ideally by each loop should correct only for disturbances that occur between it and the immediate upstream loop. In fact, in the original system each loop corrected for all upstream disturbances. This resulted in undesirable over-correction and ringing. We added MIMO (Multiple Input Multiple Output) adaptive noise cancellers to separate the signal we wish to correct from disturbances further upstream. This adaptive control improved performance in the 1992 run

  5. 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.

  6. Alien Noise Cancellation

    Indian Academy of Sciences (India)

    First page Back Continue Last page Overview Graphics. Full FEXT Cancellation. Expectation Maximization based Algorithms. Partial Cancellation. Optimal Choice of what to Cancel and what not to! Alien Noise Cancellation. Efficient Crosstalk channel estimation. In addition:

  7. An Adaptive Noise Cancellation System Based on Linear and Widely Linear Complex Valued Least Mean Square Algorithms for Removing Electrooculography Artifacts from Electroencephalography Signals

    Directory of Open Access Journals (Sweden)

    Engin Cemal MENGÜÇ

    2018-03-01

    Full Text Available In this study, an adaptive noise cancellation (ANC system based on linear and widely linear (WL complex valued least mean square (LMS algorithms is designed for removing electrooculography (EOG artifacts from electroencephalography (EEG signals. The real valued EOG and EEG signals (Fp1 and Fp2 given in dataset are primarily expressed as a complex valued signal in the complex domain. Then, using the proposed ANC system, the EOG artifacts are eliminated in the complex domain from the EEG signals. Expression of these signals in the complex domain allows us to remove EOG artifacts from two EEG channels simultaneously. Moreover, in this study, it has been shown that the complex valued EEG signal exhibits noncircular behavior, and in the case, the WL-CLMS algorithm enhances the performance of the ANC system compared to real-valued LMS and CLMS algorithms. Simulation results support the proposed approach.

  8. Adaptive noise cancellation

    International Nuclear Information System (INIS)

    Akram, N.

    1999-01-01

    In this report we describe the concept of adaptive noise canceling, an alternative method of estimating signals corrupted by additive noise of interference. The method uses 'primary' input containing the corrupted signal and a 'reference' input containing noise correlated in some unknown way with the primary noise, the reference input is adaptively filtered and subtracted from the primary input to obtain the signal estimate. Adaptive filtering before subtraction allows the treatment of inputs that are deterministic or stochastic, stationary or time variable. When the reference input is free of signal and certain other conditions are met then noise in the primary input can be essentially eliminated without signal distortion. It is further shown that the adaptive filter also acts as notch filter. Simulated results illustrate the usefulness of the adaptive noise canceling technique. (author)

  9. Non-linear signal response functions and their effects on the statistical and noise cancellation properties of isotope ratio measurements by multi-collector plasma mass spectrometry

    International Nuclear Information System (INIS)

    Doherty, W.

    2013-01-01

    A nebulizer-centric response function model of the analytical inductively coupled argon plasma ion source was used to investigate the statistical frequency distributions and noise reduction factors of simultaneously measured flicker noise limited isotope ion signals and their ratios. The response function model was extended by assuming i) a single gaussian distributed random noise source (nebulizer gas pressure fluctuations) and ii) the isotope ion signal response is a parabolic function of the nebulizer gas pressure. Model calculations of ion signal and signal ratio histograms were obtained by applying the statistical method of translation to the non-linear response function model of the plasma. Histograms of Ni, Cu, Pr, Tl and Pb isotope ion signals measured using a multi-collector plasma mass spectrometer were, without exception, negative skew. Histograms of the corresponding isotope ratios of Ni, Cu, Tl and Pb were either positive or negative skew. There was a complete agreement between the measured and model calculated histogram skew properties. The nebulizer-centric response function model was also used to investigate the effect of non-linear response functions on the effectiveness of noise cancellation by signal division. An alternative noise correction procedure suitable for parabolic signal response functions was derived and applied to measurements of isotope ratios of Cu, Ni, Pb and Tl. The largest noise reduction factors were always obtained when the non-linearity of the response functions was taken into account by the isotope ratio calculation. Possible applications of the nebulizer-centric response function model to other types of analytical instrumentation, large amplitude signal noise sources (e.g., lasers, pumped nebulizers) and analytical error in isotope ratio measurements by multi-collector plasma mass spectrometry are discussed. - Highlights: ► Isotope ion signal noise is modelled as a parabolic transform of a gaussian variable. ► Flicker

  10. Vocal Noise Cancellation From Respiratory Sounds

    National Research Council Canada - National Science Library

    Moussavi, Zahra

    2001-01-01

    Although background noise cancellation for speech or electrocardiographic recording is well established, however when the background noise contains vocal noises and the main signal is a breath sound...

  11. Noise canceling in-situ detection

    Science.gov (United States)

    Walsh, David O.

    2014-08-26

    Technologies applicable to noise canceling in-situ NMR detection and imaging are disclosed. An example noise canceling in-situ NMR detection apparatus may comprise one or more of a static magnetic field generator, an alternating magnetic field generator, an in-situ NMR detection device, an auxiliary noise detection device, and a computer.

  12. Active noise cancellation in hearing devices

    DEFF Research Database (Denmark)

    2013-01-01

    Disclosed is a hearing device system comprising at least one hearing aid circuitry and at least one active noise cancellation unit, the at least one hearing aid circuitry comprises at least one input transducer adapted to convert a first audio signal to an electric audio signal; a signal processor...... connected to the at least one input transducer and adapted to process said electric audio signal by at least partially correcting for a hearing loss of a user; an output transducer adapted to generate from at least said processed electric audio signal a sound pressure in an ear canal of the user, whereby...... the generated sound pressure is at least partially corrected for the hearing loss of the user; ; the at least one active noise cancellation unit being adapted to provide an active noise cancellation signal adapted to perform active noise cancellation of an acoustical signal entering the ear canal in addition...

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

    Science.gov (United States)

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

    2014-01-01

    infrasonic array at the Newport News-Williamsburg International Airport early in the year 2013. A pattern of pressure burst, high-coherence intervals, and diminishing-coherence intervals was observed for all takeoff and landing events without exception. The results of a phased microphone vs. linear infrasonic array comparison will be presented.

  14. Thermal Noise Canceling in LNAs : A Review

    NARCIS (Netherlands)

    Nauta, Bram; Klumperink, Eric A.M.; Bruccoleri, Frederico

    2004-01-01

    Most wide-band amplifiers suffer from a fundamental trade-off between noise figure NF and source impedance matching, which limits NF to values typically above 3dB. Recently, a feed-forward noise canceling technique has been proposed to break this trade-off. This paper reviews the principle of the

  15. Amplifiers Exploiting Thermal Noise Canceling: A Review

    NARCIS (Netherlands)

    Klumperink, Eric A.M.; Bruccoleri, F.; Stroet, P.M.; Stroet, Peter; Nauta, Bram

    2004-01-01

    Wide-band LNAs suffer from a fundamental trade-off between noise figure NF and source impedance matching, which limits NF to values typically above 3dB. Recently, a feed-forward noise canceling technique has been proposed to break this trade-off. This paper reviews the principle of the technique and

  16. Amplifiers Exploiting Thermal Noise Canceling: A Review

    OpenAIRE

    Klumperink, Eric A.M.; Bruccoleri, Federico; Stroet, Peter; Nauta, Bram

    2004-01-01

    Wide-band LNAs suffer from a fundamental trade-off between noise figure NF and source impedance matching, which limits NF to values typically above 3dB. Recently, a feed-forward noise canceling technique has been proposed to break this trade-off. This paper reviews the principle of the technique and its key properties. Although the technique has been applied to wideband CMOS LNAs, it can just as well be implemented exploiting transconductance elements realized with oth...

  17. Thermal Noise Canceling in LNAs: A Review

    OpenAIRE

    Nauta, Bram; Klumperink, Eric A.M.; Bruccoleri, Frederico

    2004-01-01

    Most wide-band amplifiers suffer from a fundamental trade-off between noise figure NF and source impedance matching, which limits NF to values typically above 3dB. Recently, a feed-forward noise canceling technique has been proposed to break this trade-off. This paper reviews the principle of the technique and its key properties. Although the technique has been applied to wideband CMOS LNAs, it can just as well be implemented exploiting transconductance elements realized with other types of t...

  18. Multireference adaptive noise canceling applied to the EEG.

    Science.gov (United States)

    James, C J; Hagan, M T; Jones, R D; Bones, P J; Carroll, G J

    1997-08-01

    The technique of multireference adaptive noise canceling (MRANC) is applied to enhance transient nonstationarities in the electroeancephalogram (EEG), with the adaptation implemented by means of a multilayer-perception artificial neural network (ANN). The method was applied to recorded EEG segments and the performance on documented nonstationarities recorded. The results show that the neural network (nonlinear) gives an improvement in performance (i.e., signal-to-noise ratio (SNR) of the nonstationarities) compared to a linear implementation of MRANC. In both cases an improvement in the SNR was obtained. The advantage of the spatial filtering aspect of MRANC is highlighted when the performance of MRANC is compared to that of the inverse auto-regressive filtering of the EEG, a purely temporal filter.

  19. 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.

  20. 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.

  1. An inductorless wideband LNA with a new noise canceling technique

    OpenAIRE

    MOGHADAM, POURIA PAZHOUHESH; ABRISHAMIFAR, ADIB

    2017-01-01

    An inductorless wideband low-noise amplifier (LNA) employing a new noise canceling technique for multistandard applications is presented. The main amplifier has a cascode common gate structure, which provides good input impedance matching and isolation. The proposed noise canceling technique not only improves the noise figure and power gain but also embeds a g$_{m}$-boosting technique in itself, which reduces the power consumption of the main amplifier. Using current-steering and ...

  2. Noise cancellation properties of displacement noise free interferometer

    International Nuclear Information System (INIS)

    Sato, Shuichi; Kawamura, Seiji; Nishizawa, Atsushi; Chen Yanbei

    2010-01-01

    We have demonstrated the practical feasibility of a displacement- and frequency-noise-free laser interferometer (DFI) by partially implementing a recently proposed optical configuration using bi-directional Mach-Zehnder interferometers (MZIs). The noise cancellation efficiency was evaluated by comparing the displacement noise spectrum of the MZIs and the DFI, demonstrating up to 50 dB of noise cancellation. In addition, the possible extension of DFI as QND device is explored.

  3. Inference from the futures: ranking the noise cancelling accuracy of realized measures

    DEFF Research Database (Denmark)

    Mirone, Giorgio

    We consider the log-linear relationship between futures contracts and their underlying assets and show that in the classical Brownian semi-martingale (BSM) framework the two series must, by no-arbitrage, have the same integrated variance. We then introduce the concept of noise cancelling...... measures in the presence of noise. Moreover, a thorough simulation analysis is employed to evaluate the estimators' sensitivity to different price and noise processes, and sampling frequencies....

  4. Development of a Voice Activity Controlled Noise Canceller

    Science.gov (United States)

    Abid Noor, Ali O.; Samad, Salina Abdul; Hussain, Aini

    2012-01-01

    In this paper, a variable threshold voice activity detector (VAD) is developed to control the operation of a two-sensor adaptive noise canceller (ANC). The VAD prohibits the reference input of the ANC from containing some strength of actual speech signal during adaptation periods. The novelty of this approach resides in using the residual output from the noise canceller to control the decisions made by the VAD. Thresholds of full-band energy and zero-crossing features are adjusted according to the residual output of the adaptive filter. Performance evaluation of the proposed approach is quoted in terms of signal to noise ratio improvements as well mean square error (MSE) convergence of the ANC. The new approach showed an improved noise cancellation performance when tested under several types of environmental noise. Furthermore, the computational power of the adaptive process is reduced since the output of the adaptive filter is efficiently calculated only during non-speech periods. PMID:22778667

  5. Development of a Voice Activity Controlled Noise Canceller

    Directory of Open Access Journals (Sweden)

    Aini Hussain

    2012-05-01

    Full Text Available In this paper, a variable threshold voice activity detector (VAD is developed to control the operation of a two-sensor adaptive noise canceller (ANC. The VAD prohibits the reference input of the ANC from containing some strength of actual speech signal during adaptation periods. The novelty of this approach resides in using the residual output from the noise canceller to control the decisions made by the VAD. Thresholds of full-band energy and zero-crossing features are adjusted according to the residual output of the adaptive filter. Performance evaluation of the proposed approach is quoted in terms of signal to noise ratio improvements as well mean square error (MSE convergence of the ANC. The new approach showed an improved noise cancellation performance when tested under several types of environmental noise. Furthermore, the computational power of the adaptive process is reduced since the output of the adaptive filter is efficiently calculated only during non-speech periods.

  6. Hardware Implementation of LMS-Based Adaptive Noise Cancellation Core with Low Resource Utilization

    Directory of Open Access Journals (Sweden)

    Omid Sharifi Tehrani

    2011-10-01

    Full Text Available A hardware implementation of adaptive noise cancellation (ANC core is proposed. Adaptive filters are widely used in different applications such as adaptive noise cancellation, prediction, equalization, inverse modeling and system identification. FIR adaptive filters are mostly used because of their low computation costs and their linear phase. Least mean squared algorithm (LMS is used to train FIR adaptive filter weights. Advances in semiconductor technology especially in digital signal processors (DSP and field programmable gate arrays (FPGA with hundreds of mega hertz in speed, will allow digital designers to embed essential digital signal processing units in small chips. But designing a synthesizable core on an FPGA is not always as simple as DSP chips due to complexity and limitations of FPGAs. In this paper we design anLMS-based FIR adaptive filter for adaptive noise cancellation based on VHDL97 hardware description language (HDL and Xilinx SPARTAN3E (XC3S500E which utilizes low resources and is high performance and FPGA-brand independent so can be implemented on different FPGA brands (Xilinx, ALTERA, ACTEL. Simulations are done in MODELSIM and MATLAB and implementation is done with Xilinx ISE. Finally, result are compared with other papers for better judgment.

  7. Laboratory Investigation of Noise-Canceling Headphones Utilizing "Mr. Blockhead"

    Science.gov (United States)

    Koser, John

    2013-01-01

    While I was co-teaching an introductory course in musical acoustics a few years ago, our class investigated several pieces of equipment designed for audio purposes. One piece of such equipment was a pair of noise-canceling headphones. Our students were curious as to how these devices were in eliminating background noise and whether they indeed…

  8. Experimental testing of the noise-canceling processor.

    Science.gov (United States)

    Collins, Michael D; Baer, Ralph N; Simpson, Harry J

    2011-09-01

    Signal-processing techniques for localizing an acoustic source buried in noise are tested in a tank experiment. Noise is generated using a discrete source, a bubble generator, and a sprinkler. The experiment has essential elements of a realistic scenario in matched-field processing, including complex source and noise time series in a waveguide with water, sediment, and multipath propagation. The noise-canceling processor is found to outperform the Bartlett processor and provide the correct source range for signal-to-noise ratios below -10 dB. The multivalued Bartlett processor is found to outperform the Bartlett processor but not the noise-canceling processor. © 2011 Acoustical Society of America

  9. Laboratory Investigation of Noise-Canceling Headphones Utilizing ``Mr. Blockhead''

    Science.gov (United States)

    Koser, John

    2013-09-01

    While I was co-teaching an introductory course in musical acoustics a few years ago, our class investigated several pieces of equipment designed for audio purposes. One piece of such equipment was a pair of noise-canceling headphones. Our students were curious as to how these devices were in eliminating background noise and whether they indeed block low-frequency sounds as advertised.

  10. A wideband Noise-Canceling CMOS LNA exploiting a transformer

    NARCIS (Netherlands)

    Blaakmeer, S.C.; Klumperink, Eric A.M.; Leenaerts, Domine M.W.; Nauta, Bram

    2006-01-01

    Abstract — A broadband LNA incorporating single-ended to differential conversion, has been successfully implemented using a noise-canceling technique and a single on-chip transformer. The LNA achieves a high voltage gain of 19dB, a wideband input match (2.5–4.0 GHz), and a Noise Figure of 4–5.4 dB,

  11. A wideband Noise-Canceling CMOS LNA exploiting a transformer

    NARCIS (Netherlands)

    Blaakmeer, S.C.; Klumperink, Eric A.M.; Leenaerts, Domine M.W.; Nauta, Bram

    2006-01-01

    A broadband LNA incorporating single-ended to differential conversion, has been successfully implemented using a noise-canceling technique and a single on-chip transformer. The LNA achieves a high voltage gain of 19dB, a wideband input match (2.5–4.0 GHz), and a Noise Figure of 4–5.4 dB, while

  12. Simulation for noise cancellation using LMS adaptive filter

    Science.gov (United States)

    Lee, Jia-Haw; Ooi, Lu-Ean; Ko, Ying-Hao; Teoh, Choe-Yung

    2017-06-01

    In this paper, the fundamental algorithm of noise cancellation, Least Mean Square (LMS) algorithm is studied and enhanced with adaptive filter. The simulation of the noise cancellation using LMS adaptive filter algorithm is developed. The noise corrupted speech signal and the engine noise signal are used as inputs for LMS adaptive filter algorithm. The filtered signal is compared to the original noise-free speech signal in order to highlight the level of attenuation of the noise signal. The result shows that the noise signal is successfully canceled by the developed adaptive filter. The difference of the noise-free speech signal and filtered signal are calculated and the outcome implies that the filtered signal is approaching the noise-free speech signal upon the adaptive filtering. The frequency range of the successfully canceled noise by the LMS adaptive filter algorithm is determined by performing Fast Fourier Transform (FFT) on the signals. The LMS adaptive filter algorithm shows significant noise cancellation at lower frequency range.

  13. Transient plasma estimation: a noise cancelling/identification approach

    International Nuclear Information System (INIS)

    Candy, J.V.; Casper, T.; Kane, R.

    1985-03-01

    The application of a noise cancelling technique to extract energy storage information from sensors occurring during fusion reactor experiments on the Tandem Mirror Experiment-Upgrade (TMX-U) at the Lawrence Livermore National Laboratory (LLNL) is examined. We show how this technique can be used to decrease the uncertainty in the corresponding sensor measurements used for diagnostics in both real-time and post-experimental environments. We analyze the performance of algorithm on the sensor data and discuss the various tradeoffs. The algorithm suggested is designed using SIG, an interactive signal processing package developed at LLNL

  14. Phase noise cancellation in polarisation-maintaining fibre links

    Science.gov (United States)

    Rauf, B.; Vélez López, M. C.; Thoumany, P.; Pizzocaro, M.; Calonico, D.

    2018-03-01

    The distribution of ultra-narrow linewidth laser radiation is an integral part of many challenging metrological applications. Changes in the optical pathlength induced by environmental disturbances compromise the stability and accuracy of optical fibre networks distributing the laser light and call for active phase noise cancellation. Here we present a laboratory scale optical (at 578 nm) fibre network featuring all polarisation maintaining fibres in a setup with low optical powers available and tracking voltage-controlled oscillators implemented. The stability and accuracy of this system reach performance levels below 1 × 10-19 after 10 000 s of averaging.

  15. A method of background noise cancellation for SQUID applications

    International Nuclear Information System (INIS)

    He, D F; Yoshizawa, M

    2003-01-01

    When superconducting quantum inference devices (SQUIDs) operate in low-cost shielding or unshielded environments, the environmental background noise should be reduced to increase the signal-to-noise ratio. In this paper we present a background noise cancellation method based on a spectral subtraction algorithm. We first measure the background noise and estimate the noise spectrum using fast Fourier transform (FFT), then we subtract the spectrum of background noise from that of the observed noisy signal and the signal can be reconstructed by inverse FFT of the subtracted spectrum. With this method, the background noise, especially stationary inferences, can be suppressed well and the signal-to-noise ratio can be increased. Using high-T C radio-frequency SQUID gradiometer and magnetometer, we have measured the magnetic field produced by a watch, which was placed 35 cm under a SQUID. After noise cancellation, the signal-to-noise ratio could be greatly increased. We also used this method to eliminate the vibration noise of a cryocooler SQUID

  16. 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.

  17. A computer simulation of an adaptive noise canceler with a single input

    Science.gov (United States)

    Albert, Stuart D.

    1991-06-01

    A description of an adaptive noise canceler using Widrows' LMS algorithm is presented. A computer simulation of canceler performance (adaptive convergence time and frequency transfer function) was written for use as a design tool. The simulations, assumptions, and input parameters are described in detail. The simulation is used in a design example to predict the performance of an adaptive noise canceler in the simultaneous presence of both strong and weak narrow-band signals (a cosited frequency hopping radio scenario). On the basis of the simulation results, it is concluded that the simulation is suitable for use as an adaptive noise canceler design tool; i.e., it can be used to evaluate the effect of design parameter changes on canceler performance.

  18. Multi-Stage Adaptive Noise Cancellation Technique for Synthetic Hard-α Inclusion

    International Nuclear Information System (INIS)

    Kim, Jae Joon

    2003-01-01

    Adaptive noise cancellation techniques are ideally suitable for reducing spatially varying noise due to the grain structure of material in ultrasonic nondestructive evaluation. Grain noises have an un-correlation property, while flaw echoes are correlated. Thus, adaptive filtering algorithms use the correlation properties of signals to enhance the signal-to-noise ratio (SNR) of the output signal. In this paper, a multi-stage adaptive noise cancellation (MANC) method using adaptive least mean square error (LMSE) filter for enhancing flaw detection in ultrasonic signals is proposed

  19. 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.

  20. Extraction of Overt Verbal Response from the Acoustic Noise in a Functional Magnetic Resonance Imaging Scan by Use of Segmented Active Noise Cancellation

    Science.gov (United States)

    Jung, Kwan-Jin; Prasad, Parikshit; Qin, Yulin; Anderson, John R.

    2013-01-01

    A method to extract the subject's overt verbal response from the obscuring acoustic noise in an fMRI scan is developed by applying active noise cancellation with a conventional MRI microphone. Since the EPI scanning and its accompanying acoustic noise in fMRI are repetitive, the acoustic noise in one time segment was used as a reference noise in suppressing the acoustic noise in subsequent segments. However, the acoustic noise from the scanner was affected by the subject's movements, so the reference noise was adaptively adjusted as the scanner's acoustic properties varied in time. This method was successfully applied to a cognitive fMRI experiment with overt verbal responses. PMID:15723385

  1. Characteristics of noise-canceling headphones to reduce the hearing hazard for MP3 users.

    Science.gov (United States)

    Liang, Maojin; Zhao, Fei; French, David; Zheng, Yiqing

    2012-06-01

    Three pairs of headphones [standard iPod ear buds and two noise-canceling headphones (NCHs)] were chosen to investigate frequency characteristics of noise reduction, together with their attenuation effects on preferred listening levels (PLLs) in the presence of various types of background noise. Twenty-six subjects with normal hearing chose their PLLs in quiet, street noise, and subway noise using the three headphones and with the noise-canceling system on/off. Both sets of NCHs reduced noise levels at mid- and high-frequencies. Further noise reductions occurred in low frequencies with the noise canceling system switched on. In street noise, both NCHs had similar noise reduction effects. In subway noise, better noise reduction effects were found in the expensive NCH and with noise-canceling on. A two way repeated measures analysis of variance showed that both listening conditions and headphone styles were significant influencing factors on the PLLs. Subjects tended to increase their PLLs as the background noise level increased. Compared with ear buds, PLLs obtained from NCHs-on in the presence of background noise were reduced up to 4 dB. Therefore, proper selection and use of NCHs appears beneficial in reducing the risk of hearing damage caused by high music listening levels in the presence of background noise.

  2. Use of active noise cancellation devices in caregivers in the intensive care unit.

    Science.gov (United States)

    Akhtar, S; Weigle, C G; Cheng, E Y; Toohill, R; Berens, R J

    2000-04-01

    Recent development of noise cancellation devices may offer relief from noise in the intensive care unit environment. This study was conducted to evaluate the effect of noise cancellation devices on subjective hearing assessment by caregivers in the intensive care units. Randomized, double-blind. Adult medical intensive care unit and pediatric intensive care unit of a teaching hospital. Caregivers of patients, including nurses, parents, respiratory therapists, and nursing assistants from a medical intensive care unit and pediatric intensive care, were enrolled in the study. Each participant was asked to wear the headphones, functional or nonfunctional noise cancellation devices, for a minimum of 30 mins. Subjective ambient noise level was assessed on a 10-point visual analog scale (VAS) before and during headphone use by each participant. Headphone comfort and the preference of the caregiver to wear the headphone were also evaluated on a 10-point VAS. Simultaneously, objective measurement of noise was done with a sound level meter using the decibel-A scale and at each of nine octave bands at each bedspace. The functional headphones significantly reduced the subjective assessment of noise by 2 (out of 10) VAS points (p noise profiles, based on decibel-A and octave band assessments. Noise cancellation devices improve subjective assessment of noise in caretakers. The benefit of these devices on hearing loss needs further evaluation in caregivers and critically ill patients.

  3. Newtonian-noise cancellation in large-scale interferometric GW detectors using seismic tiltmeters

    International Nuclear Information System (INIS)

    Harms, Jan; Venkateswara, Krishna

    2016-01-01

    The mitigation of terrestrial gravity noise, also known as Newtonian noise (NN), is one of the foremost challenges to improve low-frequency sensitivity of ground-based gravitational-wave detectors. At frequencies above 1 Hz, it is predicted that gravity noise from seismic surface Rayleigh waves is the dominant contribution to NN in surface detectors, and may still contribute significantly in future underground detectors. Noise cancellation based on a coherent estimate of NN using data from a seismometer array was proposed in the past. In this article, we propose an alternative scheme to cancel NN using a seismic tiltmeter. It is shown that even under pessimistic assumptions concerning the complexity of the seismic field, a single tiltmeter under each test mass of the detector is sufficient to achieve substantial noise cancellation. A technical tiltmeter design is presented to achieve the required sensitivity in the Newtonian-noise frequency band. (paper)

  4. Wide-band CMOS low-noise amplifier exploiting thermal noise canceling

    OpenAIRE

    Bruccoleri, F.; Klumperink, Eric A.M.; Nauta, Bram

    2004-01-01

    Known elementary wide-band amplifiers suffer from a fundamental tradeoff between noise figure (NF) and source impedance matching, which limits the NF to values typically above 3 dB. Global negative feedback can be used to break this tradeoff, however, at the price of potential instability. In contrast, this paper presents a feedforward noise-canceling technique, which allows for simultaneous noise and impedance matching, while canceling the noise and distortion contributions of the matching d...

  5. Shot-noise-limited optical Faraday polarimetry with enhanced laser noise cancelling

    International Nuclear Information System (INIS)

    Li, Jiaming; Luo, Le; Carvell, Jeff; Cheng, Ruihua; Lai, Tianshu; Wang, Zixin

    2014-01-01

    We present a shot-noise-limited measurement of optical Faraday rotations with sub-ten-nanoradian angular sensitivity. This extremely high sensitivity is achieved by using electronic laser noise cancelling and phase sensitive detection. Specially, an electronic laser noise canceller with a common mode rejection ratio of over 100 dB was designed and built for enhanced laser noise cancelling. By measuring the Faraday rotation of ambient air, we demonstrate an angular sensitivity of up to 9.0×10 −9  rad/√(Hz), which is limited only by the shot-noise of the photocurrent of the detector. To date, this is the highest angular sensitivity ever reported for Faraday polarimeters in the absence of cavity enhancement. The measured Verdet constant of ambient air, 1.93(3)×10 −9 rad/(G cm) at 633 nm wavelength, agrees extremely well with the earlier experiments using high finesse optical cavities. Further, we demonstrate the applications of this sensitive technique in materials science by measuring the Faraday effect of an ultrathin iron film

  6. Active Control of Fan Noise: Feasibility Study. Volume 3; Active Fan Noise Cancellation in the NASA Lewis Active Noise Control Fan Facility

    Science.gov (United States)

    Pla, Frederic G.; Hu, Ziqiang; Sutliff, Daniel L.

    1996-01-01

    This report describes the Active Noise Cancellation (ANC) System designed by General Electric and tested in the NASA Lewis Research Center's (LERC) 48 inch Active Noise Control Fan (ANCF). The goal of this study is to assess the feasibility of using wall mounted secondary acoustic sources and sensors within the duct of a high bypass turbofan aircraft engine for global active noise cancellation of fan tones. The GE ANC system is based on a modal control approach. A known acoustic mode propagating in the fan duct is canceled using an array of flush-mounted compact sound sources. The canceling modal signal is generated by a modal controller. Inputs to the controller are signals from a shaft encoder and from a microphone array which senses the residual acoustic mode in the duct. The key results are that the (6,0) was completely eliminated at the 920 Hz design frequency and substantially reduced elsewhere. The total tone power was reduced 6.8 dB (out of a possible 9.8 dB). Farfield reductions of 15 dB (SPL) were obtained. The (4,0) and (4,1) modes were reduced simultaneously yielding a 15 dB PWL decrease. The results indicate that global attenuation of PWL at the target frequency was obtained in the aft quadrant using an ANC actuator and sensor system totally contained within the duct. The quality of the results depended on precise mode generation. High spillover into spurious modes generated by the ANC actuator array caused less than optimum levels of PWL reduction. The variation in spillover is believed to be due to calibration procedure, but must be confirmed in subsequent tests.

  7. A multichannel nonlinear adaptive noise canceller based on generalized FLANN for fetal ECG extraction

    International Nuclear Information System (INIS)

    Ma, Yaping; Wei, Guo; Sun, Jinwei; Xiao, Yegui

    2016-01-01

    In this paper, a multichannel nonlinear adaptive noise canceller (ANC) based on the generalized functional link artificial neural network (FLANN, GFLANN) is proposed for fetal electrocardiogram (FECG) extraction. A FIR filter and a GFLANN are equipped in parallel in each reference channel to respectively approximate the linearity and nonlinearity between the maternal ECG (MECG) and the composite abdominal ECG (AECG). A fast scheme is also introduced to reduce the computational cost of the FLANN and the GFLANN. Two (2) sets of ECG time sequences, one synthetic and one real, are utilized to demonstrate the improved effectiveness of the proposed nonlinear ANC. The real dataset is derived from the Physionet non-invasive FECG database (PNIFECGDB) including 55 multichannel recordings taken from a pregnant woman. It contains two subdatasets that consist of 14 and 8 recordings, respectively, with each recording being 90 s long. Simulation results based on these two datasets reveal, on the whole, that the proposed ANC does enjoy higher capability to deal with nonlinearity between MECG and AECG as compared with previous ANCs in terms of fetal QRS (FQRS)-related statistics and morphology of the extracted FECG waveforms. In particular, for the second real subdataset, the F1-measure results produced by the PCA-based template subtraction (TS pca ) technique and six (6) single-reference channel ANCs using LMS- and RLS-based FIR filters, Volterra filter, FLANN, GFLANN, and adaptive echo state neural network (ESN a ) are 92.47%, 93.70%, 94.07%, 94.22%, 94.90%, 94.90%, and 95.46%, respectively. The same F1-measure statistical results from five (5) multi-reference channel ANCs (LMS- and RLS-based FIR filters, Volterra filter, FLANN, and GFLANN) for the second real subdataset turn out to be 94.08%, 94.29%, 94.68%, 94.91%, and 94.96%, respectively. These results indicate that the ESN a and GFLANN perform best, with the ESN a being slightly better than the GFLANN but about four times

  8. Comparative study of adaptive-noise-cancellation algorithms for intrusion detection systems

    International Nuclear Information System (INIS)

    Claassen, J.P.; Patterson, M.M.

    1981-01-01

    Some intrusion detection systems are susceptible to nonstationary noise resulting in frequent nuisance alarms and poor detection when the noise is present. Adaptive inverse filtering for single channel systems and adaptive noise cancellation for two channel systems have both demonstrated good potential in removing correlated noise components prior detection. For such noise susceptible systems the suitability of a noise reduction algorithm must be established in a trade-off study weighing algorithm complexity against performance. The performance characteristics of several distinct classes of algorithms are established through comparative computer studies using real signals. The relative merits of the different algorithms are discussed in the light of the nature of intruder and noise signals

  9. Benign paroxysmal positional vertigo after use of noise-canceling headphones.

    Science.gov (United States)

    Dan-Goor, Eric; Samra, Monica

    2012-01-01

    Benign paroxysmal positional vertigo (BPPV) is a common cause of vertigo. We describe a case of a woman presenting acutely with a severe episode of disabling positional vertigo. Although she had no known etiologic risk factors, this attack followed 12 hours of continuously wearing digital noise-canceling headphones. This is the first such reported association between BPPV and the use of this gadget. We also provide a short review of BPPV and speculate on the possible pathogenic mechanisms involved. Copyright © 2012 Elsevier Inc. All rights reserved.

  10. FPGA implementation of ICA algorithm for blind signal separation and adaptive noise canceling.

    Science.gov (United States)

    Kim, Chang-Min; Park, Hyung-Min; Kim, Taesu; Choi, Yoon-Kyung; Lee, Soo-Young

    2003-01-01

    An field programmable gate array (FPGA) implementation of independent component analysis (ICA) algorithm is reported for blind signal separation (BSS) and adaptive noise canceling (ANC) in real time. In order to provide enormous computing power for ICA-based algorithms with multipath reverberation, a special digital processor is designed and implemented in FPGA. The chip design fully utilizes modular concept and several chips may be put together for complex applications with a large number of noise sources. Experimental results with a fabricated test board are reported for ANC only, BSS only, and simultaneous ANC/BSS, which demonstrates successful speech enhancement in real environments in real time.

  11. A Tunable Low Noise Active Bandpass Filter Using a Noise Canceling Technique

    OpenAIRE

    Soltani, N.

    2016-01-01

    A monolithic tunable low noise active bandpass filter is presented in this study. Biasing voltages can control the center frequency and quality factor. By keeping the gain constant, the center frequency shift is 300 MHz. The quality factor can range from 90 to 290 at the center frequency. By using a noise cancelling circuit, noise is kept lower than 2.8 dB. The proposed filter is designed using MMIC technology with a center frequency of 2.4 GHz and a power consumption of 180 mW. ED02AH techno...

  12. Noise-cancelled, cavity-enhanced saturation laser spectroscopy for laser frequency stabilisation

    International Nuclear Information System (INIS)

    Vine, Glenn de; McClelland, David E; Gray, Malcolm B

    2006-01-01

    We employ a relatively simple experimental technique enabling mechanical-noise free, cavityenhanced spectroscopic measurements of an atomic transition and its hyperfine structure. We demonstrate this technique with the 532 nm frequency doubled output from a Nd:YAG laser and an iodine vapour cell. The resulting cavity-enhanced, noise-cancelled, iodine hyperfine error signal is used as a frequency reference with which we stabilise the frequency of the 1064nm Nd:YAG laser. Preliminary frequency stabilisation results are then presented

  13. A Tunable Low Noise Active Bandpass Filter Using a Noise Canceling Technique

    Directory of Open Access Journals (Sweden)

    N. Soltani

    2016-12-01

    Full Text Available A monolithic tunable low noise active bandpass filter is presented in this study. Biasing voltages can control the center frequency and quality factor. By keeping the gain constant, the center frequency shift is 300 MHz. The quality factor can range from 90 to 290 at the center frequency. By using a noise cancelling circuit, noise is kept lower than 2.8 dB. The proposed filter is designed using MMIC technology with a center frequency of 2.4 GHz and a power consumption of 180 mW. ED02AH technology is used to simulate the circuit elements.

  14. High performance magnetic bearings suitable for noise cancellation in permanent magnet motor driven pumps

    International Nuclear Information System (INIS)

    Zmood, R.; Cholewka, J.; Patak, C.; Feng, G.; Zhang, C.; Maleri, T.; Pinder, B.; McDonald, R.; Homrigh, J.

    1991-01-01

    Conventional pumps having external drive motors experience problems due to bearing noise. In addition failure of bearings and seals can lead to limited operational reliability and impaired integrity of these pumps. Pumps using DC brushless motors and magnetic bearings offer means of overcoming these problems. A design of a pump having a DC brushless motor and magnetic bearings with a potential for Naval applications in ships and submarines is discussed. In this paper attention is given to the selection of the magnetic bearings suitable for achieving active noise cancellation

  15. A 0.18 μm CMOS inductorless complementary-noise-canceling-LNA for TV tuner applications

    International Nuclear Information System (INIS)

    Yuan Haiquan; Lin Fujiang; Fu Zhongqian; Huang Lu

    2010-01-01

    This paper presents an inductorless complementary-noise-canceling LNA (CNCLNA) for TV tuners. The CNCLNA exploits single-to-differential topology, which consists of a common gate stage and a common source stage. The complementary topology can save power and improve the noise figure. Linearity is also enhanced by employing a multiple gated transistors technique. The chip is implemented in SMIC 0.18 μm CMOS technology. Measurement shows that the proposed CNCLNA achieves 13.5-16 dB voltage gain from 50 to 860 MHz, the noise figure is below 4.5 dB and has a minimum value of 2.9 dB, and the best P 1dB is -7.5 dBm at 860 MHz. The core consumes 6 mA current with a supply voltage of 1.8 V, while the core area is only 0.2 x 0.2 mm 2 . (semiconductor integrated circuits)

  16. An OFDM Receiver with Frequency Domain Diversity Combined Impulsive Noise Canceller for Underwater Network

    Science.gov (United States)

    Saotome, Rie; Hai, Tran Minh; Matsuda, Yasuto; Suzuki, Taisaku; Wada, Tomohisa

    2015-01-01

    In order to explore marine natural resources using remote robotic sensor or to enable rapid information exchange between ROV (remotely operated vehicles), AUV (autonomous underwater vehicle), divers, and ships, ultrasonic underwater communication systems are used. However, if the communication system is applied to rich living creature marine environment such as shallow sea, it suffers from generated Impulsive Noise so-called Shrimp Noise, which is randomly generated in time domain and seriously degrades communication performance in underwater acoustic network. With the purpose of supporting high performance underwater communication, a robust digital communication method for Impulsive Noise environments is necessary. In this paper, we propose OFDM ultrasonic communication system with diversity receiver. The main feature of the receiver is a newly proposed Frequency Domain Diversity Combined Impulsive Noise Canceller. The OFDM receiver utilizes 20–28 KHz ultrasonic channel and subcarrier spacing of 46.875 Hz (MODE3) and 93.750 Hz (MODE2) OFDM modulations. In addition, the paper shows Impulsive Noise distribution data measured at a fishing port in Okinawa and at a barge in Shizuoka prefectures and then proposed diversity OFDM transceivers architecture and experimental results are described. By the proposed Impulsive Noise Canceller, frame bit error rate has been decreased by 20–30%. PMID:26351656

  17. An OFDM Receiver with Frequency Domain Diversity Combined Impulsive Noise Canceller for Underwater Network.

    Science.gov (United States)

    Saotome, Rie; Hai, Tran Minh; Matsuda, Yasuto; Suzuki, Taisaku; Wada, Tomohisa

    2015-01-01

    In order to explore marine natural resources using remote robotic sensor or to enable rapid information exchange between ROV (remotely operated vehicles), AUV (autonomous underwater vehicle), divers, and ships, ultrasonic underwater communication systems are used. However, if the communication system is applied to rich living creature marine environment such as shallow sea, it suffers from generated Impulsive Noise so-called Shrimp Noise, which is randomly generated in time domain and seriously degrades communication performance in underwater acoustic network. With the purpose of supporting high performance underwater communication, a robust digital communication method for Impulsive Noise environments is necessary. In this paper, we propose OFDM ultrasonic communication system with diversity receiver. The main feature of the receiver is a newly proposed Frequency Domain Diversity Combined Impulsive Noise Canceller. The OFDM receiver utilizes 20-28 KHz ultrasonic channel and subcarrier spacing of 46.875 Hz (MODE3) and 93.750 Hz (MODE2) OFDM modulations. In addition, the paper shows Impulsive Noise distribution data measured at a fishing port in Okinawa and at a barge in Shizuoka prefectures and then proposed diversity OFDM transceivers architecture and experimental results are described. By the proposed Impulsive Noise Canceller, frame bit error rate has been decreased by 20-30%.

  18. An OFDM Receiver with Frequency Domain Diversity Combined Impulsive Noise Canceller for Underwater Network

    Directory of Open Access Journals (Sweden)

    Rie Saotome

    2015-01-01

    Full Text Available In order to explore marine natural resources using remote robotic sensor or to enable rapid information exchange between ROV (remotely operated vehicles, AUV (autonomous underwater vehicle, divers, and ships, ultrasonic underwater communication systems are used. However, if the communication system is applied to rich living creature marine environment such as shallow sea, it suffers from generated Impulsive Noise so-called Shrimp Noise, which is randomly generated in time domain and seriously degrades communication performance in underwater acoustic network. With the purpose of supporting high performance underwater communication, a robust digital communication method for Impulsive Noise environments is necessary. In this paper, we propose OFDM ultrasonic communication system with diversity receiver. The main feature of the receiver is a newly proposed Frequency Domain Diversity Combined Impulsive Noise Canceller. The OFDM receiver utilizes 20–28 KHz ultrasonic channel and subcarrier spacing of 46.875 Hz (MODE3 and 93.750 Hz (MODE2 OFDM modulations. In addition, the paper shows Impulsive Noise distribution data measured at a fishing port in Okinawa and at a barge in Shizuoka prefectures and then proposed diversity OFDM transceivers architecture and experimental results are described. By the proposed Impulsive Noise Canceller, frame bit error rate has been decreased by 20–30%.

  19. Study of Noise Canceling Performance of Feedforward Fuzzy-Based ANC System under Non-Causal Condition

    DEFF Research Database (Denmark)

    Mojallali, Hamed; Izadi-Zamanabadi, Roozbeh; Amini, Rouzbeh

    of noise canceling performance of feed-forward fuzzy-based ANC systems for ducts under non-causal condition is presented. For this purpose, we use fuzzy filtered-x algorithm as an adaptive filter and the results are compared with classical filteredx algorithm which is employed under the same conditions......Feed-forward active noise control (ANC) systems act as adaptive systems to control and cancel undesired signals and noises. If the delay in the noise canceling subsystems increases more than the delays in the primary path, non-causal condition will occur in these systems. In this paper, study....... Analysis shows that ANC systems using fuzzy algorithm has better efficiency for noise cancellation in non-causal condition....

  20. Adaptive HIFU noise cancellation for simultaneous therapy and imaging using an integrated HIFU/imaging transducer

    International Nuclear Information System (INIS)

    Jeong, Jong Seob; Cannata, Jonathan Matthew; Shung, K Kirk

    2010-01-01

    It was previously demonstrated that it is feasible to simultaneously perform ultrasound therapy and imaging of a coagulated lesion during treatment with an integrated transducer that is capable of high intensity focused ultrasound (HIFU) and B-mode ultrasound imaging. It was found that coded excitation and fixed notch filtering upon reception could significantly reduce interference caused by the therapeutic transducer. During HIFU sonication, the imaging signal generated with coded excitation and fixed notch filtering had a range side-lobe level of less than -40 dB, while traditional short-pulse excitation and fixed notch filtering produced a range side-lobe level of -20 dB. The shortcoming is, however, that relatively complicated electronics may be needed to utilize coded excitation in an array imaging system. It is for this reason that in this paper an adaptive noise canceling technique is proposed to improve image quality by minimizing not only the therapeutic interference, but also the remnant side-lobe 'ripples' when using the traditional short-pulse excitation. The performance of this technique was verified through simulation and experiments using a prototype integrated HIFU/imaging transducer. Although it is known that the remnant ripples are related to the notch attenuation value of the fixed notch filter, in reality, it is difficult to find the optimal notch attenuation value due to the change in targets or the media resulted from motion or different acoustic properties even during one sonication pulse. In contrast, the proposed adaptive noise canceling technique is capable of optimally minimizing both the therapeutic interference and residual ripples without such constraints. The prototype integrated HIFU/imaging transducer is composed of three rectangular elements. The 6 MHz center element is used for imaging and the outer two identical 4 MHz elements work together to transmit the HIFU beam. Two HIFU elements of 14.4 mm x 20.0 mm dimensions could

  1. Adaptive HIFU noise cancellation for simultaneous therapy and imaging using an integrated HIFU/imaging transducer.

    Science.gov (United States)

    Jeong, Jong Seob; Cannata, Jonathan Matthew; Shung, K Kirk

    2010-04-07

    It was previously demonstrated that it is feasible to simultaneously perform ultrasound therapy and imaging of a coagulated lesion during treatment with an integrated transducer that is capable of high intensity focused ultrasound (HIFU) and B-mode ultrasound imaging. It was found that coded excitation and fixed notch filtering upon reception could significantly reduce interference caused by the therapeutic transducer. During HIFU sonication, the imaging signal generated with coded excitation and fixed notch filtering had a range side-lobe level of less than -40 dB, while traditional short-pulse excitation and fixed notch filtering produced a range side-lobe level of -20 dB. The shortcoming is, however, that relatively complicated electronics may be needed to utilize coded excitation in an array imaging system. It is for this reason that in this paper an adaptive noise canceling technique is proposed to improve image quality by minimizing not only the therapeutic interference, but also the remnant side-lobe 'ripples' when using the traditional short-pulse excitation. The performance of this technique was verified through simulation and experiments using a prototype integrated HIFU/imaging transducer. Although it is known that the remnant ripples are related to the notch attenuation value of the fixed notch filter, in reality, it is difficult to find the optimal notch attenuation value due to the change in targets or the media resulted from motion or different acoustic properties even during one sonication pulse. In contrast, the proposed adaptive noise canceling technique is capable of optimally minimizing both the therapeutic interference and residual ripples without such constraints. The prototype integrated HIFU/imaging transducer is composed of three rectangular elements. The 6 MHz center element is used for imaging and the outer two identical 4 MHz elements work together to transmit the HIFU beam. Two HIFU elements of 14.4 mm x 20.0 mm dimensions could

  2. Controlling kilometre-scale interferometric detectors for gravitational wave astronomy: Active phase noise cancellation using EOMs

    International Nuclear Information System (INIS)

    Arnaud, N.; Balembois, L.; Bizouard, M.A.; Brisson, V.; Casanueva, J.; Cavalier, F.; Davier, M.; Frey, V.; Hello, P.; Huet, D.; Leroy, N.; Loriette, V.; Maksimovic, I.; Robinet, F.

    2017-01-01

    The second generation of Gravitational waves detectors are kilometric Michelson interferometers with additional recycling Fabry–Perot cavities on the arms and ​the addition of two more recycling cavities to enhance their sensitivity, with the particularity that all the mirrors are suspended. In order to control them a new technique, based on the use of auxiliary lasers, has been developed to bring the interferometer to its working point, with all the cavities on their resonance, in an adiabatic way. The implementation of this technique in Advanced Virgo is under preparation and the propagation of a stable laser through a 3-km optical fibre is one of the most problematic issues. A new technique of active phase noise cancellation based on the use of Electro Optical Modulators has been developed, and a first prototype has been successfully tested.

  3. System and method for motor fault detection using stator current noise cancellation

    Science.gov (United States)

    Zhou, Wei; Lu, Bin; Nowak, Michael P.; Dimino, Steven A.

    2010-12-07

    A system and method for detecting incipient mechanical motor faults by way of current noise cancellation is disclosed. The system includes a controller configured to detect indicia of incipient mechanical motor faults. The controller further includes a processor programmed to receive a baseline set of current data from an operating motor and define a noise component in the baseline set of current data. The processor is also programmed to acquire at least on additional set of real-time operating current data from the motor during operation, redefine the noise component present in each additional set of real-time operating current data, and remove the noise component from the operating current data in real-time to isolate any fault components present in the operating current data. The processor is then programmed to generate a fault index for the operating current data based on any isolated fault components.

  4. System and method for bearing fault detection using stator current noise cancellation

    Science.gov (United States)

    Zhou, Wei; Lu, Bin; Habetler, Thomas G.; Harley, Ronald G.; Theisen, Peter J.

    2010-08-17

    A system and method for detecting incipient mechanical motor faults by way of current noise cancellation is disclosed. The system includes a controller configured to detect indicia of incipient mechanical motor faults. The controller further includes a processor programmed to receive a baseline set of current data from an operating motor and define a noise component in the baseline set of current data. The processor is also programmed to repeatedly receive real-time operating current data from the operating motor and remove the noise component from the operating current data in real-time to isolate any fault components present in the operating current data. The processor is then programmed to generate a fault index for the operating current data based on any isolated fault components.

  5. Controlling kilometre-scale interferometric detectors for gravitational wave astronomy: Active phase noise cancellation using EOMs

    Energy Technology Data Exchange (ETDEWEB)

    Arnaud, N.; Balembois, L.; Bizouard, M.A.; Brisson, V. [LAL, Univ. Paris-Sud, IN2P3/CNRS, Univ. Paris-Saclay, Orsay (France); Casanueva, J., E-mail: casanuev@lal.in2p3.fr [LAL, Univ. Paris-Sud, IN2P3/CNRS, Univ. Paris-Saclay, Orsay (France); Cavalier, F.; Davier, M.; Frey, V.; Hello, P.; Huet, D.; Leroy, N. [LAL, Univ. Paris-Sud, IN2P3/CNRS, Univ. Paris-Saclay, Orsay (France); Loriette, V.; Maksimovic, I. [ESPCI, CNRS, F-75005 Paris (France); Robinet, F. [LAL, Univ. Paris-Sud, IN2P3/CNRS, Univ. Paris-Saclay, Orsay (France)

    2017-02-11

    The second generation of Gravitational waves detectors are kilometric Michelson interferometers with additional recycling Fabry–Perot cavities on the arms and ​the addition of two more recycling cavities to enhance their sensitivity, with the particularity that all the mirrors are suspended. In order to control them a new technique, based on the use of auxiliary lasers, has been developed to bring the interferometer to its working point, with all the cavities on their resonance, in an adiabatic way. The implementation of this technique in Advanced Virgo is under preparation and the propagation of a stable laser through a 3-km optical fibre is one of the most problematic issues. A new technique of active phase noise cancellation based on the use of Electro Optical Modulators has been developed, and a first prototype has been successfully tested.

  6. Towards a first design of a Newtonian-noise cancellation system for Advanced LIGO

    International Nuclear Information System (INIS)

    Coughlin, M; Mukund, N; Mitra, S; Harms, J; Driggers, J; Adhikari, R

    2016-01-01

    Newtonian gravitational noise from seismic fields is predicted to be a limiting noise source at low frequency for second generation gravitational-wave detectors. Mitigation of this noise will be achieved by Wiener filtering using arrays of seismometers deployed in the vicinity of all test masses. In this work, we present optimized configurations of seismometer arrays using a variety of simplified models of the seismic field based on seismic observations at LIGO Hanford. The model that best fits the seismic measurements leads to noise reduction limited predominantly by seismometer self-noise. A first simplified design of seismic arrays for Newtonian-noise cancellation at the LIGO sites is presented, which suggests that it will be sufficient to monitor surface displacement inside the buildings. (paper)

  7. Controlling kilometre-scale interferometric detectors for gravitational wave astronomy: Active phase noise cancellation using EOMs

    Science.gov (United States)

    Arnaud, N.; Balembois, L.; Bizouard, M. A.; Brisson, V.; Casanueva, J.; Cavalier, F.; Davier, M.; Frey, V.; Hello, P.; Huet, D.; Leroy, N.; Loriette, V.; Maksimovic, I.; Robinet, F.

    2017-02-01

    The second generation of Gravitational waves detectors are kilometric Michelson interferometers with additional recycling Fabry-Perot cavities on the arms and ​the addition of two more recycling cavities to enhance their sensitivity, with the particularity that all the mirrors are suspended. In order to control them a new technique, based on the use of auxiliary lasers, has been developed to bring the interferometer to its working point, with all the cavities on their resonance, in an adiabatic way. The implementation of this technique in Advanced Virgo is under preparation and the propagation of a stable laser through a 3-km optical fibre is one of the most problematic issues. A new technique of active phase noise cancellation based on the use of Electro Optical Modulators has been developed, and a first prototype has been successfully tested.

  8. A single-ended CMOS sensing circuit for MEMS gyroscope with noise cancellation

    KAUST Repository

    Elsayed, Mohannad Yomn

    2010-06-01

    In this work, a complete single-ended readout circuit for capacitive MEMS gyroscope using chopper stabilization technique is presented. A novel noise cancellation technique is used to get rid of the bias noise. The circuit offers superior performance over state of the art readout circuits in terms of cost, gain, and noise for the given area and power consumption. The full circuit exhibits a gain of 58dB, a power dissipation of 1.3mW and an input referred noise of 12nV/√Hz. This would significantly improve the overall sensitivity of the gyroscope. The full circuit has been fabricated in 0.6um CMOS technology and it occupies an area of 0.4mm × 1mm. © 2010 IEEE.

  9. A single-ended CMOS sensing circuit for MEMS gyroscope with noise cancellation

    KAUST Repository

    Elsayed, Mohannad Yomn; Emira, Ahmed; Sedky, Sherif M.; Habib, S. E. D.

    2010-01-01

    In this work, a complete single-ended readout circuit for capacitive MEMS gyroscope using chopper stabilization technique is presented. A novel noise cancellation technique is used to get rid of the bias noise. The circuit offers superior performance over state of the art readout circuits in terms of cost, gain, and noise for the given area and power consumption. The full circuit exhibits a gain of 58dB, a power dissipation of 1.3mW and an input referred noise of 12nV/√Hz. This would significantly improve the overall sensitivity of the gyroscope. The full circuit has been fabricated in 0.6um CMOS technology and it occupies an area of 0.4mm × 1mm. © 2010 IEEE.

  10. Transform Domain Robust Variable Step Size Griffiths' Adaptive Algorithm for Noise Cancellation in ECG

    Science.gov (United States)

    Hegde, Veena; Deekshit, Ravishankar; Satyanarayana, P. S.

    2011-12-01

    The electrocardiogram (ECG) is widely used for diagnosis of heart diseases. Good quality of ECG is utilized by physicians for interpretation and identification of physiological and pathological phenomena. However, in real situations, ECG recordings are often corrupted by artifacts or noise. Noise severely limits the utility of the recorded ECG and thus needs to be removed, for better clinical evaluation. In the present paper a new noise cancellation technique is proposed for removal of random noise like muscle artifact from ECG signal. A transform domain robust variable step size Griffiths' LMS algorithm (TVGLMS) is proposed for noise cancellation. For the TVGLMS, the robust variable step size has been achieved by using the Griffiths' gradient which uses cross-correlation between the desired signal contaminated with observation or random noise and the input. The algorithm is discrete cosine transform (DCT) based and uses symmetric property of the signal to represent the signal in frequency domain with lesser number of frequency coefficients when compared to that of discrete Fourier transform (DFT). The algorithm is implemented for adaptive line enhancer (ALE) filter which extracts the ECG signal in a noisy environment using LMS filter adaptation. The proposed algorithm is found to have better convergence error/misadjustment when compared to that of ordinary transform domain LMS (TLMS) algorithm, both in the presence of white/colored observation noise. The reduction in convergence error achieved by the new algorithm with desired signal decomposition is found to be lower than that obtained without decomposition. The experimental results indicate that the proposed method is better than traditional adaptive filter using LMS algorithm in the aspects of retaining geometrical characteristics of ECG signal.

  11. 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...

  12. Recognition of Voice Commands by Multisource ASR and Noise Cancellation in a Smart Home Environment

    OpenAIRE

    Vacher , Michel; Lecouteux , Benjamin; Portet , François

    2012-01-01

    International audience; In this paper, we present a multisource ASR system to detect home automation orders in various everyday listening conditions in a realistic home. The system is based on a state of the art echo cancellation stage that feeds recently introduced ASR techniques. The evaluation was conducted on a realistic noisy data set acquired in a smart home where a microphone was placed near the noise source and several other microphones were placed in different rooms. This distant spe...

  13. Reduced Pain and Anxiety with Music and Noise-Canceling Headphones During Shockwave Lithotripsy.

    Science.gov (United States)

    Karalar, Mustafa; Keles, Ibrahim; Doğantekin, Engin; Kahveci, Orhan Kemal; Sarici, Hasmet

    2016-06-01

    We assessed the effects of music and noise-canceling headphones (NCHs) on perceived patient pain and anxiety from extracorporeal shockwave lithotripsy (SWL). Patients with renal calculi scheduled for SWL were prospectively enrolled. All 89 patients between the ages of 19 and 80 years were informed about this study and then randomized into three groups: Group 1 (controls), no headphones and music; Group 2, music with NCHs (patients listened to Turkish classical music with NCHs during SWL); and Group 3, music with non-NCHs (patients listened to Turkish classical music with non-NCHs during SWL). Hemodynamic and respiratory parameters were recorded before and just after the SWL session. All patient visual analog scale (VAS) and State-Trait Anxiety Inventory (STAI) scores were recorded just after the SWL procedure. There were significant differences in VAS scores among the groups (5.1, 3.6, and 4.5, respectively, p < 0.001), including between Groups 2 and 3 (p = 0.018). There were also significant differences in STAI-State anxiety scores among the groups (43.1, 33.5, and 38.9, respectively, p = 0.001), including between Groups 2 and 3 (p = 0.04). Music therapy during SWL reduced pain and anxiety. Music therapy with NCHs was more effective for pain and anxiety reduction. To reduce pain and anxiety, nonpharmacologic therapies such as music therapy with NCHs during SWL should be investigated further and used routinely.

  14. 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.

  15. Active noise cancellation of low frequency noise propagating in a duct

    Directory of Open Access Journals (Sweden)

    Farhad Forouharmajd

    2012-01-01

    Conclusions: With regard to the wide range of frequencies of different noise sources, having optimized circumstances in the duct, microphone location on the duct body or even the distance of the speakers may be important in signal processing, noise sampling and anti noise production.

  16. Beat Noise Cancellation in 2-D Optical Code-Division Multiple-Access Systems Using Optical Hard-Limiter Array

    Science.gov (United States)

    Dang, Ngoc T.; Pham, Anh T.; Cheng, Zixue

    We analyze the beat noise cancellation in two-dimensional optical code-division multiple-access (2-D OCDMA) systems using an optical hard-limiter (OHL) array. The Gaussian shape of optical pulse is assumed and the impact of pulse propagation is considered. We also take into account the receiver noise and multiple access interference (MAI) in the analysis. The numerical results show that, when OHL array is employed, the system performance is greatly improved compared with the cases without OHL array. Also, parameters needed for practical system design are comprehensively analyzed.

  17. A Hybrid Semi-Digital Transimpedance Amplifier With Noise Cancellation Technique for Nanopore-Based DNA Sequencing.

    Science.gov (United States)

    Hsu, Chung-Lun; Jiang, Haowei; Venkatesh, A G; Hall, Drew A

    2015-10-01

    Over the past two decades, nanopores have been a promising technology for next generation deoxyribonucleic acid (DNA) sequencing. Here, we present a hybrid semi-digital transimpedance amplifier (HSD-TIA) to sense the minute current signatures introduced by single-stranded DNA (ssDNA) translocating through a nanopore, while discharging the baseline current using a semi-digital feedback loop. The amplifier achieves fast settling by adaptively tuning a DC compensation current when a step input is detected. A noise cancellation technique reduces the total input-referred current noise caused by the parasitic input capacitance. Measurement results show the performance of the amplifier with 31.6 M Ω mid-band gain, 950 kHz bandwidth, and 8.5 fA/ √Hz input-referred current noise, a 2× noise reduction due to the noise cancellation technique. The settling response is demonstrated by observing the insertion of a protein nanopore in a lipid bilayer. Using the nanopore, the HSD-TIA was able to measure ssDNA translocation events.

  18. Carbon Nanotube Thin Films for Active Noise Cancellation, Solar Energy Harvesting, and Energy Storage in Building Windows

    Science.gov (United States)

    Hu, Shan

    This research explores the application of carbon nanotube (CNT) films for active noise cancellation, solar energy harvesting and energy storage in building windows. The CNT-based components developed herein can be integrated into a solar-powered active noise control system for a building window. First, the use of a transparent acoustic transducer as both an invisible speaker for auxiliary audio playback and for active noise cancellation is accomplished in this work. Several challenges related to active noise cancellation in the window are addressed. These include secondary path estimation and directional cancellation of noise so as to preserve auxiliary audio and internal sounds while preventing transmission of external noise into the building. Solar energy can be harvested at a low rate of power over long durations while acoustic sound cancellation requires short durations of high power. A supercapacitor based energy storage system is therefore considered for the window. Using CNTs as electrode materials, two generations of flexible, thin, and fully solid-state supercapacitors are developed that can be integrated into the window frame. Both generations consist of carbon nanotube films coated on supporting substrates as electrodes and a solid-state polymer gel layer for the electrolyte. The first generation is a single-cell parallel-plate supercapacitor with a working voltage of 3 Volts. Its energy density is competitive with commercially available supercapacitors (which use liquid electrolyte). For many applications that will require higher working voltage, the second-generation multi-cell supercapacitor is developed. A six-cell device with a working voltage as high as 12 Volts is demonstrated here. Unlike the first generation's 3D structure, the second generation has a novel planar (2D) architecture, which makes it easy to integrate multiple cells into a thin and flexible supercapacitor. The multi-cell planar supercapacitor has energy density exceeding that of

  19. Joint Use of Adaptive Equalization and Cyclic Noise Cancellation for Band-Limited OQAM Based Multi-Carrier Transmission in Power-Line Communication Systems

    Science.gov (United States)

    Kunishima, Hiromitsu; Koga, Hisao; Muta, Osamu; Akaiwa, Yoshihiko

    Power-line communication (PLC) technique is one method to realize high-speed communications in a home network. In PLC channels, the transmission signal quality is degraded by colored non-Gaussian noise as well as frequency-selectivity of the channels. In this paper, we describe our investigation of the performance of a OQAM-MCT system in which a noise canceller is used jointly with a time-domain per-subcarrier adaptive equalizer. Furthermore, we propose a noise cancellation method designed for the OQAM-MCT system. The performance of the OQAM-MCT system is evaluated in PLC channels with measured impulse responses in the presence of measured noise. Computer simulation results show that the bit rate of the OQAM-MCT system is improved using both an adaptive equalizer and noise canceller, and that the OQAM-MCT system achieves better performance than an OFDM system with an insufficient length of the guard interval.

  20. A Small-Area Self-Biased Wideband CMOS Balun LNA with Noise Cancelling and Gain Enhancement

    DEFF Research Database (Denmark)

    Bruun, Erik; Andreani, Pietro; Custódio, J. R.

    2010-01-01

    In this paper we present a low-power and small-area balun LNA. The proposed inverter-based topology uses selfbiasing and noise cancelling, yielding a very robust LNA with a low NF. Comparing this circuit with a conventional inverterbased circuit, we obtain a ∼3 dB enhancement in voltage gain......, with improved robustness against PVT variations. Simulations results in a 130 nm CMOS technology show a 17.7dB voltage gain, nearly flat over a wide bandwidth (200MHz-1GHz), and an NF of approximately 4dB. The total power consumption is below 7.5 mW, with a very small die area of 0.007 mm2. All data...

  1. Force sensing based on coherent quantum noise cancellation in a hybrid optomechanical cavity with squeezed-vacuum injection

    International Nuclear Information System (INIS)

    Motazedifard, Ali; Bemani, F; Naderi, M H; Roknizadeh, R; Vitali, D

    2016-01-01

    We propose and analyse a feasible experimental scheme for a quantum force sensor based on the elimination of backaction noise through coherent quantum noise cancellation (CQNC) in a hybrid atom-cavity optomechanical setup assisted with squeezed vacuum injection. The force detector, which allows for a continuous, broadband detection of weak forces well below the standard quantum limit (SQL), is formed by a single optical cavity simultaneously coupled to a mechanical oscillator and to an ensemble of ultracold atoms. The latter acts as a negative-mass oscillator so that atomic noise exactly cancels the backaction noise from the mechanical oscillator due to destructive quantum interference. Squeezed vacuum injection enforces this cancellation and allows sub-SQL sensitivity to be reached in a very wide frequency band, and at much lower input laser powers. (paper)

  2. The effect of losses on the quantum-noise cancellation in the SU(1,1) interferometer

    International Nuclear Information System (INIS)

    Xin, Jun; Wang, Hailong; Jing, Jietai

    2016-01-01

    Quantum-noise cancellation (QNC) is an effective method to control the noise of the quantum system, which reduces or even eliminates the noise of the quantum systems by utilizing destructive interference in the quantum system. However, QNC can be extremely dependent on the losses inside the system. In this letter, we experimentally and theoretically study how the losses can affect the QNC in the SU(1,1) interferometer. We find that losses in the different arms inside the SU(1,1) interferometer can have different effects on the QNC in the output fields from the SU(1,1) interferometer. And the QNC in the SU(1,1) interferometer can almost be insensitive to the losses in some cases. Our findings may find its potential applications in the quantum noise control.

  3. Force sensing based on coherent quantum noise cancellation in a hybrid optomechanical cavity with squeezed-vacuum injection

    Science.gov (United States)

    Motazedifard, Ali; Bemani, F.; Naderi, M. H.; Roknizadeh, R.; Vitali, D.

    2016-07-01

    We propose and analyse a feasible experimental scheme for a quantum force sensor based on the elimination of backaction noise through coherent quantum noise cancellation (CQNC) in a hybrid atom-cavity optomechanical setup assisted with squeezed vacuum injection. The force detector, which allows for a continuous, broadband detection of weak forces well below the standard quantum limit (SQL), is formed by a single optical cavity simultaneously coupled to a mechanical oscillator and to an ensemble of ultracold atoms. The latter acts as a negative-mass oscillator so that atomic noise exactly cancels the backaction noise from the mechanical oscillator due to destructive quantum interference. Squeezed vacuum injection enforces this cancellation and allows sub-SQL sensitivity to be reached in a very wide frequency band, and at much lower input laser powers.

  4. The effect of losses on the quantum-noise cancellation in the SU(1,1) interferometer

    Energy Technology Data Exchange (ETDEWEB)

    Xin, Jun; Wang, Hailong [State Key Laboratory of Precision Spectroscopy, School of Physics and Materials Science, East China Normal University, Shanghai 200062 (China); Jing, Jietai, E-mail: jtjing@phy.ecnu.edu.cn [State Key Laboratory of Precision Spectroscopy, School of Physics and Materials Science, East China Normal University, Shanghai 200062 (China); Collaborative Innovation Center of Extreme Optics, Shanxi University, Taiyuan, Shanxi 030006 (China)

    2016-08-01

    Quantum-noise cancellation (QNC) is an effective method to control the noise of the quantum system, which reduces or even eliminates the noise of the quantum systems by utilizing destructive interference in the quantum system. However, QNC can be extremely dependent on the losses inside the system. In this letter, we experimentally and theoretically study how the losses can affect the QNC in the SU(1,1) interferometer. We find that losses in the different arms inside the SU(1,1) interferometer can have different effects on the QNC in the output fields from the SU(1,1) interferometer. And the QNC in the SU(1,1) interferometer can almost be insensitive to the losses in some cases. Our findings may find its potential applications in the quantum noise control.

  5. An Embedded, Eight Channel, Noise Canceling, Wireless, Wearable sEMG Data Acquisition System With Adaptive Muscle Contraction Detection.

    Science.gov (United States)

    Ergeneci, Mert; Gokcesu, Kaan; Ertan, Erhan; Kosmas, Panagiotis

    2018-02-01

    Wearable technology has gained increasing popularity in the applications of healthcare, sports science, and biomedical engineering in recent years. Because of its convenient nature, the wearable technology is particularly useful in the acquisition of the physiological signals. Specifically, the (surface electromyography) sEMG systems, which measure the muscle activation potentials, greatly benefit from this technology in both clinical and industrial applications. However, the current wearable sEMG systems have several drawbacks including inefficient noise cancellation, insufficient measurement quality, and difficult integration to customized applications. Additionally, none of these sEMG data acquisition systems can detect sEMG signals (i.e., contractions), which provides a valuable environment for further studies such as human machine interaction, gesture recognition, and fatigue tracking. To this end, we introduce an embedded, eight channel, noise canceling, wireless, wearable sEMG data acquisition system with adaptive muscle contraction detection. Our design consists of two stages, which are the sEMG sensors and the multichannel data acquisition unit. For the first stage, we propose a low cost, dry, and active sEMG sensor that captures the muscle activation potentials, a data acquisition unit that evaluates these captured multichannel sEMG signals and transmits them to a user interface. In the data acquisition unit, the sEMG signals are processed through embedded, adaptive methods in order to reject the power line noise and detect the muscle contractions. Through extensive experiments, we demonstrate that our sEMG sensor outperforms a widely used commercially available product and our data acquisition system achieves 4.583 dB SNR gain with accuracy in the detection of the contractions.

  6. A wide bandwidth fractional-N synthesizer for LTE with phase noise cancellation using a hybrid-ΔΣ-DAC and charge re-timing

    NARCIS (Netherlands)

    Ye, D.; Lu, Ping; Andreani, Pietro; van der Zee, Ronan A.R.

    2013-01-01

    This paper presents a 1MHz bandwidth, ΔΣ fractional-N PLL as the frequency synthesizer for LTE. A noise cancellation path composed of a novel hybrid ΔΣ DAC with 9 output bits is incorporated into the PLL in order to cancel the out-of-band phase noise caused by the quantization error. Further, a

  7. Reducing the Effects of Background Noise during Auditory Functional Magnetic Resonance Imaging of Speech Processing: Qualitative and Quantitative Comparisons between Two Image Acquisition Schemes and Noise Cancellation

    Science.gov (United States)

    Blackman, Graham A.; Hall, Deborah A.

    2011-01-01

    Purpose: The intense sound generated during functional magnetic resonance imaging (fMRI) complicates studies of speech and hearing. This experiment evaluated the benefits of using active noise cancellation (ANC), which attenuates the level of the scanner sound at the participant's ear by up to 35 dB around the peak at 600 Hz. Method: Speech and…

  8. The effect on recognition memory of noise cancelling headphones in a noisy environment with native and nonnative speakers

    Directory of Open Access Journals (Sweden)

    Brett R C Molesworth

    2014-01-01

    Full Text Available Noise has the potential to impair cognitive performance. For nonnative speakers, the effect of noise on performance is more severe than their native counterparts. What remains unknown is the effectiveness of countermeasures such as noise attenuating devices in such circumstances. Therefore, the main aim of the present research was to examine the effectiveness of active noise attenuating countermeasures in the presence of simulated aircraft noise for both native and nonnative English speakers. Thirty-two participants, half native English speakers and half native German speakers completed four recognition (cued recall tasks presented in English under four different audio conditions, all in the presence of simulated aircraft noise. The results of the research indicated that in simulated aircraft noise at 65 dB(A, performance of nonnative English speakers was poorer than for native English speakers. The beneficial effects of noise cancelling headphones in improving the signal to noise ratio led to an improved performance for nonnative speakers. These results have particular importance for organizations operating in a safety-critical environment such as aviation.

  9. A comparative evaluation of adaptive noise cancellation algorithms for minimizing motion artifacts in a forehead-mounted wearable pulse oximeter.

    Science.gov (United States)

    Comtois, Gary; Mendelson, Yitzhak; Ramuka, Piyush

    2007-01-01

    Wearable physiological monitoring using a pulse oximeter would enable field medics to monitor multiple injuries simultaneously, thereby prioritizing medical intervention when resources are limited. However, a primary factor limiting the accuracy of pulse oximetry is poor signal-to-noise ratio since photoplethysmographic (PPG) signals, from which arterial oxygen saturation (SpO2) and heart rate (HR) measurements are derived, are compromised by movement artifacts. This study was undertaken to quantify SpO2 and HR errors induced by certain motion artifacts utilizing accelerometry-based adaptive noise cancellation (ANC). Since the fingers are generally more vulnerable to motion artifacts, measurements were performed using a custom forehead-mounted wearable pulse oximeter developed for real-time remote physiological monitoring and triage applications. This study revealed that processing motion-corrupted PPG signals by least mean squares (LMS) and recursive least squares (RLS) algorithms can be effective to reduce SpO2 and HR errors during jogging, but the degree of improvement depends on filter order. Although both algorithms produced similar improvements, implementing the adaptive LMS algorithm is advantageous since it requires significantly less operations.

  10. Time Delay Mechanical-noise Cancellation (TDMC) to Provide Order of Magnitude Improvements in Radio Science Observations

    Science.gov (United States)

    Atkinson, D. H.; Babuscia, A.; Lazio, J.; Asmar, S.

    2017-12-01

    Many Radio Science investigations, including the determinations of planetary masses, measurements of planetary atmospheres, studies of the solar wind, and solar system tests of relativistic gravity, rely heavily on precision Doppler tracking. Recent and currently proposed missions such as VERITAS, Bepi Colombo, Juno have shown that the largest error source in the precision Doppler tracking data is noise in the Doppler system. This noise is attributed to un-modeled motions of the ground antenna's phase center and is commonly referred to as "antenna mechanical noise." Attempting to reduce this mechanical noise has proven difficult since the deep space communications antennas utilize large steel structures that are already optimized for mechanical stability. Armstrong et al. (2008) have demonstrated the Time Delay Mechanical-noise Cancellation (TDMC) concept using Goldstone DSN antennas (70 m & 34 m) and the Cassinispacecraft to show that the mechanical noise of the 70 m antenna could be suppressed when two-way Doppler tracking from the 70 m antenna and the receive-only Doppler data from the smaller, stiffer 34 m antenna were combined with suitable delays. The proof-of-concept confirmed that the mechanical noise in the final Doppler observable was reduced to that of the stiffer, more stable antenna. Caltech's Owens Valley Radio Observatory (OVRO) near Bishop, CA now has six 10.4 m diameter antennas, a consequence of the closure of Combined Array for Research in Millimeter Astronomy (CARMA). In principle, a 10 m antenna can lead to an order-of-magnitude improvement for the mechanical noise correction, as the smaller dish offers better mechanical stability compared to a DSN 34-m antenna. These antennas also have existing Ka-band receiving systems, and preliminary discussions with the OVRO staff suggest that much of the existing signal path could be used for Radio Science observations.

  11. Design of circular differential microphone arrays

    CERN Document Server

    Benesty, Jacob; Cohen, Israel

    2015-01-01

    Recently, we proposed a completely novel and efficient way to design differential beamforming algorithms for linear microphone arrays. Thanks to this very flexible approach, any order of differential arrays can be designed. Moreover, they can be made robust against white noise amplification, which is the main inconvenience in these types of arrays. The other well-known problem with linear arrays is that electronic steering is not really feasible.  In this book, we extend all these fundamental ideas to circular microphone arrays and show that we can design small and compact differential arrays of any order that can be electronically steered in many different directions and offer a good degree of control of the white noise amplification problem, high directional gain, and frequency-independent response. We also present a number of practical examples, demonstrating that differential beamforming with circular microphone arrays is likely one of the best candidates for applications involving speech enhancement (i....

  12. Adaptive noise cancellation

    International Nuclear Information System (INIS)

    Rizwan, N.

    1999-01-01

    Wavelet analysis consists of decomposing a signal or an image into a hierarchical set of approximations and details. The levels in the hierarchy correspond to those in a dyadic scale. Wavelet provide an alternative to classical Short Time Fourier Transforms for the analysis of non-stationary signals. Wavelets are defined in continuous time and discrete time. Recently Discrete Wavelet Transform (DWT) had emerged as a popular technique in Image Compression. DWT has high decorrelation and energy compaction efficiency. In this report, the effect of level of decomposition on image compression was studied and results are compared with DCT based image compression. DWT proved better in compression as there was high energy compaction and compressed image was free from blocking artifacts. (author)

  13. Linewidth-tolerant 10-Gbit/s 16-QAM transmission using a pilot-carrier based phase-noise cancelling technique.

    Science.gov (United States)

    Nakamura, Moriya; Kamio, Yukiyoshi; Miyazaki, Tetsuya

    2008-07-07

    We experimentally demonstrated linewidth-tolerant 10-Gbit/s (2.5-Gsymbol/s) 16-quadrature amplitude modulation (QAM) by using a distributed-feedback laser diode (DFB-LD) with a linewidth of 30 MHz. Error-free operation, a bit-error rate (BER) of noise canceling capability provided by a pilot-carrier and standard electronic pre-equalization to suppress inter-symbol interference (ISI) gave clear 16-QAM constellations and floor-less BER characteristics. We evaluated the BER characteristics by real-time measurement of six (three different thresholds for each I- and Q-component) symbol error rates (SERs) with simultaneous constellation observation.

  14. 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)

  15. 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.

  16. A wideband LNA employing gate-inductive-peaking and noise-canceling techniques in 0.18 μm CMOS

    International Nuclear Information System (INIS)

    Bao Kuan; Fan Xiangning; Li Wei; Zhang Li; Wang Zhigong

    2012-01-01

    This paper presents a wideband low noise amplifier (LNA) for multi-standard radio applications. The low noise characteristic is achieved by the noise-canceling technique while the bandwidth is enhanced by gate-inductive-peaking technique. High-frequency noise performance is consequently improved by the flattened gain over the entire operating frequency band. Fabricated in 0.18 μm CMOS process, the LNA achieves 2.5 GHz of −3 dB bandwidth and 16 dB of gain. The gain variation is within ±0.8 dB from 300 MHz to 2.2 GHz. The measured noise figure (NF) and average IIP3 are 3.4 dB and −2 dBm, respectively. The proposed LNA occupies 0.39 mm 2 core chip area. Operating at 1.8 V, the LNA drains a current of 11.7 mA. (semiconductor integrated circuits)

  17. 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.

  18. 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

  19. 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....

  20. 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...

  1. 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

  2. A wideband high-linearity RF receiver front-end in CMOS

    NARCIS (Netherlands)

    Arkesteijn, V.J.; Klumperink, Eric A.M.; Nauta, Bram

    This paper presents a wideband high-linearity RF receiver-front-end, implemented in standard 0.18 μm CMOS technology. The design employs a noise-canceling LNA in combination with two passive mixers, followed by lowpass-filtering and amplification at IF. The achieved bandwidth is >2 GHz, with a noise

  3. 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

  4. 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

  5. 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

  6. 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

  7. 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

  8. 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

  9. 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...

  10. 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.

  11. Theory and applications of spherical microphone array processing

    CERN Document Server

    Jarrett, Daniel P; Naylor, Patrick A

    2017-01-01

    This book presents the signal processing algorithms that have been developed to process the signals acquired by a spherical microphone array. Spherical microphone arrays can be used to capture the sound field in three dimensions and have received significant interest from researchers and audio engineers. Algorithms for spherical array processing are different to corresponding algorithms already known in the literature of linear and planar arrays because the spherical geometry can be exploited to great beneficial effect. The authors aim to advance the field of spherical array processing by helping those new to the field to study it efficiently and from a single source, as well as by offering a way for more experienced researchers and engineers to consolidate their understanding, adding either or both of breadth and depth. The level of the presentation corresponds to graduate studies at MSc and PhD level. This book begins with a presentation of some of the essential mathematical and physical theory relevant to ...

  12. 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.)

  13. 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.

  14. 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

  15. Study and Design of Differential Microphone Arrays

    CERN Document Server

    Benesty, Jacob

    2013-01-01

    Microphone arrays have attracted a lot of interest over the last few decades since they have the potential to solve many important problems such as noise reduction/speech enhancement, source separation, dereverberation, spatial sound recording, and source localization/tracking, to name a few. However, the design and implementation of microphone arrays with beamforming algorithms is not a trivial task when it comes to processing broadband signals such as speech. Indeed, in most sensor arrangements, the beamformer tends to have a frequency-dependent response. One exception, perhaps, is the family of differential microphone arrays (DMAs) that have the promise to form frequency-independent responses. Moreover, they have the potential to attain high directional gains with small and compact apertures. As a result, this type of microphone arrays has drawn much research and development attention recently. This book is intended to provide a systematic study of DMAs from a signal processing perspective. The primary obj...

  16. 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

  17. 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.

  18. Improved Design of Microphone-Array Hearing Aids

    National Research Council Canada - National Science Library

    Greenberg, Julie

    1994-01-01

    ...). Research on microphone array hearing aids is motivated by the lack of success of single-microphone systems, as well as the documented advantages of binaural hearing and multiple-element sensing systems...

  19. Locating noise sources with a microphone array

    International Nuclear Information System (INIS)

    Bale, A.; Johnson, D.

    2010-01-01

    Noise pollution is one of the contributors to the public opposition of wind farms. Most of the noise produced by turbines is caused by the aerodynamic interactions between the turbine blades and the surrounding air. This poster presentation discussed a series of aeroacoustic tests conducted to account for the different in vortical structures caused by the rotation of the blades. Microphone arrays were used measure and locate the source of noise. A beam forming technique was used to measure the noise using an algorithm that identified a scanning grid on a plane where the source was thought to be located. It delayed each microphone's signal by the length of time required for the sound to travel from the scan position to each microphone, and accounted for the amplitudes according to the distance from the scan position to each microphone. Demonstration test cases were conducted using piezo buzzers attached to aluminum bars and mounted to the shaft of a DC motor that produced a rotational diameter of 0.95 meter. The buzzers were placed 1 meter from the array. Multiple sound sources at the same frequency were identified, and the moving sources were accurately measured and located. tabs., figs.

  20. 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.)

  1. 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...

  2. 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

  3. 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....

  4. 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....

  5. 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)

  6. 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.

  7. 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.

  8. 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.

  9. 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...

  10. 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

  11. 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...

  12. 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...

  13. 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

  14. 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...

  15. 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...

  16. 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.

  17. 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...

  18. 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.

  19. 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...

  20. 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...

  1. 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.

  2. 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

  3. 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...

  4. 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.

  5. 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...

  6. 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.

  7. 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

  8. Acoustic Source Localization in Aircraft Interiors Using Microphone Array Technologies

    Science.gov (United States)

    Sklanka, Bernard J.; Tuss, Joel R.; Buehrle, Ralph D.; Klos, Jacob; Williams, Earl G.; Valdivia, Nicolas

    2006-01-01

    Using three microphone array configurations at two aircraft body stations on a Boeing 777-300ER flight test, the acoustic radiation characteristics of the sidewall and outboard floor system are investigated by experimental measurement. Analysis of the experimental data is performed using sound intensity calculations for closely spaced microphones, PATCH Inverse Boundary Element Nearfield Acoustic Holography, and Spherical Nearfield Acoustic Holography. Each method is compared assessing strengths and weaknesses, evaluating source identification capability for both broadband and narrowband sources, evaluating sources during transient and steady-state conditions, and quantifying field reconstruction continuity using multiple array positions.

  9. 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.

  10. 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...

  11. Acoustooptic linear algebra processors - Architectures, algorithms, and applications

    Science.gov (United States)

    Casasent, D.

    1984-01-01

    Architectures, algorithms, and applications for systolic processors are described with attention to the realization of parallel algorithms on various optical systolic array processors. Systolic processors for matrices with special structure and matrices of general structure, and the realization of matrix-vector, matrix-matrix, and triple-matrix products and such architectures are described. Parallel algorithms for direct and indirect solutions to systems of linear algebraic equations and their implementation on optical systolic processors are detailed with attention to the pipelining and flow of data and operations. Parallel algorithms and their optical realization for LU and QR matrix decomposition are specifically detailed. These represent the fundamental operations necessary in the implementation of least squares, eigenvalue, and SVD solutions. Specific applications (e.g., the solution of partial differential equations, adaptive noise cancellation, and optimal control) are described to typify the use of matrix processors in modern advanced signal processing.

  12. 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.

  13. 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.

  14. 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.

  15. 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...

  16. 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...

  17. 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...

  18. 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.

  19. 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

  20. 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…

  1. 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...

  2. 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

  3. 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

  4. 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.

  5. 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...

  6. 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.

  7. 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.

  8. 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...

  9. 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.

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

    Science.gov (United States)

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

    2002-05-01

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

  11. 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....

  12. Noise Reduction with Microphone Arrays for Speaker Identification

    Energy Technology Data Exchange (ETDEWEB)

    Cohen, Z

    2011-12-22

    Reducing acoustic noise in audio recordings is an ongoing problem that plagues many applications. This noise is hard to reduce because of interfering sources and non-stationary behavior of the overall background noise. Many single channel noise reduction algorithms exist but are limited in that the more the noise is reduced; the more the signal of interest is distorted due to the fact that the signal and noise overlap in frequency. Specifically acoustic background noise causes problems in the area of speaker identification. Recording a speaker in the presence of acoustic noise ultimately limits the performance and confidence of speaker identification algorithms. In situations where it is impossible to control the environment where the speech sample is taken, noise reduction filtering algorithms need to be developed to clean the recorded speech of background noise. Because single channel noise reduction algorithms would distort the speech signal, the overall challenge of this project was to see if spatial information provided by microphone arrays could be exploited to aid in speaker identification. The goals are: (1) Test the feasibility of using microphone arrays to reduce background noise in speech recordings; (2) Characterize and compare different multichannel noise reduction algorithms; (3) Provide recommendations for using these multichannel algorithms; and (4) Ultimately answer the question - Can the use of microphone arrays aid in speaker identification?

  13. 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

  14. 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.

  15. 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...

  16. 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...

  17. 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.

  18. 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.

  19. 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...

  20. 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....

  1. 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...

  2. 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

  3. 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.

  4. Wideband Low Noise Amplifiers Exploiting Thermal Noise Cancellation

    NARCIS (Netherlands)

    Bruccoleri, F.; Klumperink, Eric A.M.; Nauta, Bram

    2005-01-01

    Low Noise Amplifiers (LNAs) are commonly used to amplify signals that are too weak for direct processing for example in radio or cable receivers. Traditionally, low noise amplifiers are implemented via tuned amplifiers, exploiting inductors and capacitors in resonating LC-circuits. This can render

  5. Noise cancellation in magnetoencephalography and electroencephalography with isolated reference sensors

    Science.gov (United States)

    Kraus, Jr., Robert H.; Espy, Michelle A.; Matlachov, Andrei; Volegov, Petr

    2010-06-01

    An apparatus measures electromagnetic signals from a weak signal source. A plurality of primary sensors is placed in functional proximity to the weak signal source with an electromagnetic field isolation surface arranged adjacent the primary sensors and between the weak signal source and sources of ambient noise. A plurality of reference sensors is placed adjacent the electromagnetic field isolation surface and arranged between the electromagnetic isolation surface and sources of ambient noise.

  6. Newtonian noise cancellation in tensor gravitational wave detector

    International Nuclear Information System (INIS)

    Paik, Ho Jung; Harms, Jan

    2016-01-01

    Terrestrial gravity noise produced by ambient seismic and infrasound fields poses one of the main sensitivity limitations in low-frequency ground-based gravitational-wave (GW) detectors. This noise needs to be suppressed by 3-5 orders of magnitude in the frequency band 10 mHz to 1 Hz, which is extremely challenging. We present a new approach that greatly facilitates cancellation of gravity noise in full-tensor GW detectors. It makes explicit use of the direction of propagation of a GW, and can therefore either be implemented in directional searches for GWs or in observations of known sources. We show that suppression of the Newtonian-noise foreground is greatly facilitated using the extra strain channels in full-tensor GW detectors. Only a modest number of auxiliary, high-sensitivity environmental sensors is required to achieve noise suppression by a few orders of magnitude. (paper)

  7. An improved VSS NLMS algorithm for active noise cancellation

    Science.gov (United States)

    Sun, Yunzhuo; Wang, Mingjiang; Han, Yufei; Zhang, Congyan

    2017-08-01

    In this paper, an improved variable step size NLMS algorithm is proposed. NLMS has fast convergence rate and low steady state error compared to other traditional adaptive filtering algorithm. But there is a contradiction between the convergence speed and steady state error that affect the performance of the NLMS algorithm. Now, we propose a new variable step size NLMS algorithm. It dynamically changes the step size according to current error and iteration times. The proposed algorithm has simple formulation and easily setting parameters, and effectively solves the contradiction in NLMS. The simulation results show that the proposed algorithm has a good tracking ability, fast convergence rate and low steady state error simultaneously.

  8. An improved affine projection algorithm for active noise cancellation

    Science.gov (United States)

    Zhang, Congyan; Wang, Mingjiang; Han, Yufei; Sun, Yunzhuo

    2017-08-01

    Affine projection algorithm is a signal reuse algorithm, and it has a good convergence rate compared to other traditional adaptive filtering algorithm. There are two factors that affect the performance of the algorithm, which are step factor and the projection length. In the paper, we propose a new variable step size affine projection algorithm (VSS-APA). It dynamically changes the step size according to certain rules, so that it can get smaller steady-state error and faster convergence speed. Simulation results can prove that its performance is superior to the traditional affine projection algorithm and in the active noise control (ANC) applications, the new algorithm can get very good results.

  9. Adaptive Noise Canceling Menggunakan Algoritma Least Mean Square (Lms)

    OpenAIRE

    Nardiana, Anita; Sumaryono, Sari Sujoko

    2011-01-01

    Noise is inevitable in communication system. In some cases, noise can disturb signal. It is veryannoying as the received signal is jumbled with the noise itself. To reduce or remove noise, filter lowpass,highpass or bandpass can solve the problems, but this method cannot reach a maximum standard. One ofthe alternatives to solve the problem is by using adaptive filter. Adaptive algorithm frequently used is LeastMean Square (LMS) Algorithm which is compatible to Finite Impulse Response (FIR). T...

  10. Adaptive noise canceling of electrocardiogram artifacts in single channel electroencephalogram.

    Science.gov (United States)

    Cho, Sung Pil; Song, Mi Hye; Park, Young Cheol; Choi, Ho Seon; Lee, Kyoung Joung

    2007-01-01

    A new method for estimating and eliminating electrocardiogram (ECG) artifacts from single channel scalp electroencephalogram (EEG) is proposed. The proposed method consists of emphasis of QRS complex from EEG using least squares acceleration (LSA) filter, generation of synchronized pulse with R-peak and ECG artifacts estimation and elimination using adaptive filter. The performance of the proposed method was evaluated using simulated and real EEG recordings, we found that the ECG artifacts were successfully estimated and eliminated in comparison with the conventional multi-channel techniques, which are independent component analysis (ICA) and ensemble average (EA) method. From this we can conclude that the proposed method is useful for the detecting and eliminating the ECG artifacts from single channel EEG and simple to use for ambulatory/portable EEG monitoring system.

  11. Adaptive Beamforming Algorithms for Tow Ship Noise Canceling

    NARCIS (Netherlands)

    Robert, M.K.; Beerens, S.P.

    2002-01-01

    In towed array sonar, the directional noise originating from the tow ship, mainly machinery and hydrodynamic noise, often limits the sonar performance. When processed with classical beamforming techniques, loud tow ship noise induces high sidelobes that may hide detection of quiet targets in forward

  12. Tuning Out the World with Noise-Canceling Headphones

    Science.gov (United States)

    McCulloch, Allison W.; Whitehead, Ashley; Lovett, Jennifer N.; Whitley, Blake

    2017-01-01

    Context is what makes mathematical modeling tasks different from more traditional textbook word problems. Math problems are sometimes stripped of context as they are worked on. For modeling problems, however, context is important for making sense of the mathematics. The task should be brought back to its real-world context as often as possible. In…

  13. 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...

  14. [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.

  15. 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.

  16. 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).

  17. Robustness of a Mixed-Order Ambisonics Microphone Array for Sound Field Reproduction

    DEFF Research Database (Denmark)

    Marschall, Marton; Favrot, Sylvain Emmanuel; Buchholz, Jörg

    2012-01-01

    Spherical microphone arrays can be used to capture and reproduce the spatial characteristics of acoustic scenes. A mixed-order Ambisonics (MOA) approach was recently proposed to improve the horizontal spatial resolution of microphone arrays with a given number of transducers. In this paper...

  18. 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...

  19. 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...

  20. 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.

  1. Precision Measurements of Wind Turbine Noise using a Large Aperture Microphone Array

    DEFF Research Database (Denmark)

    Bradley, Stuart; Mikkelsen, Torben Krogh; Hünerbein, Sabine Von

    2016-01-01

    Experiments are described with a large microphone array (40 m scale) recording wind turbine noise. The array comprised 42 purpose-designed low-noise microphones simultaneously sampled at 20 kHz. Very high quality, fast, meteorological profile data was available from nearby 80 m masts and from the...

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

    Science.gov (United States)

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

    2015-11-01

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

  3. 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...

  4. 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...

  5. A Framework for Speech Enhancement with Ad Hoc Microphone Arrays

    DEFF Research Database (Denmark)

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

    2016-01-01

    Speech enhancement is vital for improved listening practices. Ad hoc microphone arrays are promising assets for this purpose. Most well-established enhancement techniques with conventional arrays can be adapted into ad hoc scenarios. Despite recent efforts to introduce various ad hoc speech...... enhancement apparatus, a common framework for integration of conventional methods into this new scheme is still missing. This paper establishes such an abstraction based on inter and intra sub-array speech coherencies. Along with measures for signal quality at the input of sub-arrays, a measure of coherency...... is proposed both for sub-array selection in local enhancement approaches, and also for selecting a proper global reference when more than one sub-array are used. Proposed methods within this framework are evaluated with regard to quantitative and qualitative measures, including array gains, the speech...

  6. 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...

  7. 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.

  8. 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.

  9. 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.

  10. 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.

  11. 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.

  12. 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

  13. 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.

  14. 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.)

  15. 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.)

  16. 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.

  17. Design of UAV-Embedded Microphone Array System for Sound Source Localization in Outdoor Environments

    Directory of Open Access Journals (Sweden)

    Kotaro Hoshiba

    2017-11-01

    Full Text Available In search and rescue activities, unmanned aerial vehicles (UAV should exploit sound information to compensate for poor visual information. This paper describes the design and implementation of a UAV-embedded microphone array system for sound source localization in outdoor environments. Four critical development problems included water-resistance of the microphone array, efficiency in assembling, reliability of wireless communication, and sufficiency of visualization tools for operators. To solve these problems, we developed a spherical microphone array system (SMAS consisting of a microphone array, a stable wireless network communication system, and intuitive visualization tools. The performance of SMAS was evaluated with simulated data and a demonstration in the field. Results confirmed that the SMAS provides highly accurate localization, water resistance, prompt assembly, stable wireless communication, and intuitive information for observers and operators.

  18. 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)

  19. 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...

  20. Design of UAV-Embedded Microphone Array System for Sound Source Localization in Outdoor Environments †

    Science.gov (United States)

    Hoshiba, Kotaro; Washizaki, Kai; Wakabayashi, Mizuho; Ishiki, Takahiro; Bando, Yoshiaki; Gabriel, Daniel; Nakadai, Kazuhiro; Okuno, Hiroshi G.

    2017-01-01

    In search and rescue activities, unmanned aerial vehicles (UAV) should exploit sound information to compensate for poor visual information. This paper describes the design and implementation of a UAV-embedded microphone array system for sound source localization in outdoor environments. Four critical development problems included water-resistance of the microphone array, efficiency in assembling, reliability of wireless communication, and sufficiency of visualization tools for operators. To solve these problems, we developed a spherical microphone array system (SMAS) consisting of a microphone array, a stable wireless network communication system, and intuitive visualization tools. The performance of SMAS was evaluated with simulated data and a demonstration in the field. Results confirmed that the SMAS provides highly accurate localization, water resistance, prompt assembly, stable wireless communication, and intuitive information for observers and operators. PMID:29099790

  1. Design of UAV-Embedded Microphone Array System for Sound Source Localization in Outdoor Environments.

    Science.gov (United States)

    Hoshiba, Kotaro; Washizaki, Kai; Wakabayashi, Mizuho; Ishiki, Takahiro; Kumon, Makoto; Bando, Yoshiaki; Gabriel, Daniel; Nakadai, Kazuhiro; Okuno, Hiroshi G

    2017-11-03

    In search and rescue activities, unmanned aerial vehicles (UAV) should exploit sound information to compensate for poor visual information. This paper describes the design and implementation of a UAV-embedded microphone array system for sound source localization in outdoor environments. Four critical development problems included water-resistance of the microphone array, efficiency in assembling, reliability of wireless communication, and sufficiency of visualization tools for operators. To solve these problems, we developed a spherical microphone array system (SMAS) consisting of a microphone array, a stable wireless network communication system, and intuitive visualization tools. The performance of SMAS was evaluated with simulated data and a demonstration in the field. Results confirmed that the SMAS provides highly accurate localization, water resistance, prompt assembly, stable wireless communication, and intuitive information for observers and operators.

  2. 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....

  3. 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.

  4. 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.

  5. 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.

  6. 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.

  7. 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

  8. 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....

  9. 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.

  10. 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.

  11. The group delay and suppression pattern of the cochlear microphonic potential recorded at the round window.

    Directory of Open Access Journals (Sweden)

    Wenxuan He

    Full Text Available BACKGROUND: It is commonly assumed that the cochlear microphonic potential (CM recorded from the round window (RW is generated at the cochlear base. Based on this assumption, the low-frequency RW CM has been measured for evaluating the integrity of mechanoelectrical transduction of outer hair cells at the cochlear base and for studying sound propagation inside the cochlea. However, the group delay and the origin of the low-frequency RW CM have not been demonstrated experimentally. METHODOLOGY/PRINCIPAL FINDINGS: This study quantified the intra-cochlear group delay of the RW CM by measuring RW CM and vibrations at the stapes and basilar membrane in gerbils. At low sound levels, the RW CM showed a significant group delay and a nonlinear growth at frequencies below 2 kHz. However, at high sound levels or at frequencies above 2 kHz, the RW CM magnitude increased proportionally with sound pressure, and the CM phase in respect to the stapes showed no significant group delay. After the local application of tetrodotoxin the RW CM below 2 kHz became linear and showed a negligible group delay. In contrast to RW CM phase, the BM vibration measured at location ∼2.5 mm from the base showed high sensitivity, sharp tuning, and nonlinearity with a frequency-dependent group delay. At low or intermediate sound levels, low-frequency RW CMs were suppressed by an additional tone near the probe-tone frequency while, at high sound levels, they were partially suppressed only at high frequencies. CONCLUSIONS/SIGNIFICANCE: We conclude that the group delay of the RW CM provides no temporal information on the wave propagation inside the cochlea, and that significant group delay of low-frequency CMs results from the auditory nerve neurophonic potential. Suppression data demonstrate that the generation site of the low-frequency RW CM shifts from apex to base as the probe-tone level increases.

  12. Resonance Control for Future Linear Accelerators

    Energy Technology Data Exchange (ETDEWEB)

    Schappert, Warren [Fermilab

    2017-05-01

    Many of the next generation of particle accelerators (LCLS II, PIP II) are designed for relatively low beam loading. Low beam loading requirement means the cavities can operate with narrow bandwidths, minimizing capital and base operational costs of the RF power system. With such narrow bandwidths, however, cavity detuning from microphonics or dynamic Lorentz Force Detuning becomes a significant factor, and in some cases can significantly increase both the acquisition cost and the operational cost of the machine. In addition to the efforts to passive environmental detuning reduction (microphonics) active resonance control for the SRF cavities for next generation linear machine will be required. State of the art in the field of the SRF Cavity active resonance control and the results from the recent efforts at FNAL will be presented in this talk.

  13. 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...

  14. 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.

  15. 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

  16. Beamforming with a circular array of microphones mounted on a rigid sphere (L)

    DEFF Research Database (Denmark)

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

    2011-01-01

    Beamforming with uniform circular microphone arrays can be used for localizing sound sources over 360. Typically, the array microphones are suspended in free space or they are mounted on a solid cylinder. However, the cylinder is often considered to be infinitely long because the scattering problem...... has no exact solution for a finite cylinder. Alternatively one can use a solid sphere. This investigation compares the performance of a circular array mounded on a rigid sphere with that of such an array in free space and mounted on an infinite cylinder, using computer simulations. The examined...

  17. 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...

  18. 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...

  19. 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,

  20. 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.

  1. 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

  2. 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.

  3. Temperature compensated, humidity insensitive, high-Tg 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...

  4. 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

  5. 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

  6. 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…

  7. 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...

  8. 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

  9. 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…

  10. Evaluation of Methods for In-Situ Calibration of Field-Deployable Microphone Phased Arrays

    Science.gov (United States)

    Humphreys, William M.; Lockard, David P.; Khorrami, Mehdi R.; Culliton, William G.; McSwain, Robert G.

    2017-01-01

    Current field-deployable microphone phased arrays for aeroacoustic flight testing require the placement of hundreds of individual sensors over a large area. Depending on the duration of the test campaign, the microphones may be required to stay deployed at the testing site for weeks or even months. This presents a challenge in regards to tracking the response (i.e., sensitivity) of the individual sensors as a function of time in order to evaluate the health of the array. To address this challenge, two different methods for in-situ tracking of microphone responses are described. The first relies on the use of an aerial sound source attached as a payload on a hovering small Unmanned Aerial System (sUAS) vehicle. The second relies on the use of individually excited ground-based sound sources strategically placed throughout the array pattern. Testing of the two methods was performed in microphone array deployments conducted at Fort A.P. Hill in 2015 and at Edwards Air Force Base in 2016. The results indicate that the drift in individual sensor responses can be tracked reasonably well using both methods. Thus, in-situ response tracking methods are useful as a diagnostic tool for monitoring the health of a phased array during long duration deployments.

  11. 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...

  12. 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...

  13. 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...

  14. 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...

  15. 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...

  16. 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

  17. 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

  18. 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.

  19. 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.

  20. 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.

  1. 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)

  2. 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...

  3. 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.

  4. 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.

  5. 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

  6. 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.

  7. 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.

  8. 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.

  9. 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.

  10. 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.

  11. 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.

  12. 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.

  13. 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.

  14. A wideband large dynamic range and high linearity RF front-end for U-band mobile DTV

    International Nuclear Information System (INIS)

    Liu Rongjiang; Liu Shengyou; Guo Guiliang; Cheng Xu; Yan Yuepeng

    2013-01-01

    A wideband large dynamic range and high linearity U-band RF front-end for mobile DTV is introduced, and includes a noise-cancelling low-noise amplifier (LNA), an RF programmable gain amplifier (RFPGA) and a current communicating passive mixer. The noise/distortion cancelling structure and RC post-distortion compensation are employed to improve the linearity of the LNA. An RFPGA with five stages provides large dynamic range and fine gain resolution. A simple resistor voltage network in the passive mixer decreases the gate bias voltage of the mixing transistor, and optimum linearity and symmetrical mixing is obtained at the same time. The RF front-end is implemented in a 0.25 μm CMOS process. Tests show that it achieves an IIP3 (third-order intercept point) of −17 dBm, a conversion gain of 39 dB, and a noise figure of 5.8 dB. The RFPGA achieves a dynamic range of −36.2 to 23.5 dB with a resolution of 0.32 dB. (semiconductor integrated circuits)

  15. 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.

  16. Improving the iterative Linear Interaction Energy approach using automated recognition of configurational transitions.

    Science.gov (United States)

    Vosmeer, C Ruben; Kooi, Derk P; Capoferri, Luigi; Terpstra, Margreet M; Vermeulen, Nico P E; Geerke, Daan P

    2016-01-01

    Recently an iterative method was proposed to enhance the accuracy and efficiency of ligand-protein binding affinity prediction through linear interaction energy (LIE) theory. For ligand binding to flexible Cytochrome P450s (CYPs), this method was shown to decrease the root-mean-square error and standard deviation of error prediction by combining interaction energies of simulations starting from different conformations. Thereby, different parts of protein-ligand conformational space are sampled in parallel simulations. The iterative LIE framework relies on the assumption that separate simulations explore different local parts of phase space, and do not show transitions to other parts of configurational space that are already covered in parallel simulations. In this work, a method is proposed to (automatically) detect such transitions during the simulations that are performed to construct LIE models and to predict binding affinities. Using noise-canceling techniques and splines to fit time series of the raw data for the interaction energies, transitions during simulation between different parts of phase space are identified. Boolean selection criteria are then applied to determine which parts of the interaction energy trajectories are to be used as input for the LIE calculations. Here we show that this filtering approach benefits the predictive quality of our previous CYP 2D6-aryloxypropanolamine LIE model. In addition, an analysis is performed of the gain in computational efficiency that can be obtained from monitoring simulations using the proposed filtering method and by prematurely terminating simulations accordingly.

  17. Linear algebra

    CERN Document Server

    Shilov, Georgi E

    1977-01-01

    Covers determinants, linear spaces, systems of linear equations, linear functions of a vector argument, coordinate transformations, the canonical form of the matrix of a linear operator, bilinear and quadratic forms, Euclidean spaces, unitary spaces, quadratic forms in Euclidean and unitary spaces, finite-dimensional space. Problems with hints and answers.

  18. 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

  19. 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.

  20. 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.

  1. 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...

  2. 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...

  3. 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...

  4. 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...

  5. 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.

  6. 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...

  7. 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...

  8. 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...

  9. 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....

  10. 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.

  11. 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

  12. 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...

  13. 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.

  14. 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.

  15. 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.

  16. 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.

  17. Linear gate

    International Nuclear Information System (INIS)

    Suwono.

    1978-01-01

    A linear gate providing a variable gate duration from 0,40μsec to 4μsec was developed. The electronic circuity consists of a linear circuit and an enable circuit. The input signal can be either unipolar or bipolar. If the input signal is bipolar, the negative portion will be filtered. The operation of the linear gate is controlled by the application of a positive enable pulse. (author)

  18. Linear Accelerators

    International Nuclear Information System (INIS)

    Vretenar, M

    2014-01-01

    The main features of radio-frequency linear accelerators are introduced, reviewing the different types of accelerating structures and presenting the main characteristics aspects of linac beam dynamics

  19. Linearization Method and Linear Complexity

    Science.gov (United States)

    Tanaka, Hidema

    We focus on the relationship between the linearization method and linear complexity and show that the linearization method is another effective technique for calculating linear complexity. We analyze its effectiveness by comparing with the logic circuit method. We compare the relevant conditions and necessary computational cost with those of the Berlekamp-Massey algorithm and the Games-Chan algorithm. The significant property of a linearization method is that it needs no output sequence from a pseudo-random number generator (PRNG) because it calculates linear complexity using the algebraic expression of its algorithm. When a PRNG has n [bit] stages (registers or internal states), the necessary computational cost is smaller than O(2n). On the other hand, the Berlekamp-Massey algorithm needs O(N2) where N(≅2n) denotes period. Since existing methods calculate using the output sequence, an initial value of PRNG influences a resultant value of linear complexity. Therefore, a linear complexity is generally given as an estimate value. On the other hand, a linearization method calculates from an algorithm of PRNG, it can determine the lower bound of linear complexity.

  20. 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.

  1. 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,

  2. 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)

  3. 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.

  4. 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...

  5. 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...

  6. Linear algebra

    CERN Document Server

    Said-Houari, Belkacem

    2017-01-01

    This self-contained, clearly written textbook on linear algebra is easily accessible for students. It begins with the simple linear equation and generalizes several notions from this equation for the system of linear equations and introduces the main ideas using matrices. It then offers a detailed chapter on determinants and introduces the main ideas with detailed proofs. The third chapter introduces the Euclidean spaces using very simple geometric ideas and discusses various major inequalities and identities. These ideas offer a solid basis for understanding general Hilbert spaces in functional analysis. The following two chapters address general vector spaces, including some rigorous proofs to all the main results, and linear transformation: areas that are ignored or are poorly explained in many textbooks. Chapter 6 introduces the idea of matrices using linear transformation, which is easier to understand than the usual theory of matrices approach. The final two chapters are more advanced, introducing t...

  7. 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

  8. 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

  9. 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.

  10. 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.

  11. 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.

  12. 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...

  13. 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.

  14. 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

  15. 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.

  16. Linear algebra

    CERN Document Server

    Stoll, R R

    1968-01-01

    Linear Algebra is intended to be used as a text for a one-semester course in linear algebra at the undergraduate level. The treatment of the subject will be both useful to students of mathematics and those interested primarily in applications of the theory. The major prerequisite for mastering the material is the readiness of the student to reason abstractly. Specifically, this calls for an understanding of the fact that axioms are assumptions and that theorems are logical consequences of one or more axioms. Familiarity with calculus and linear differential equations is required for understand

  17. Linear programming

    CERN Document Server

    Solow, Daniel

    2014-01-01

    This text covers the basic theory and computation for a first course in linear programming, including substantial material on mathematical proof techniques and sophisticated computation methods. Includes Appendix on using Excel. 1984 edition.

  18. Linear algebra

    CERN Document Server

    Liesen, Jörg

    2015-01-01

    This self-contained textbook takes a matrix-oriented approach to linear algebra and presents a complete theory, including all details and proofs, culminating in the Jordan canonical form and its proof. Throughout the development, the applicability of the results is highlighted. Additionally, the book presents special topics from applied linear algebra including matrix functions, the singular value decomposition, the Kronecker product and linear matrix equations. The matrix-oriented approach to linear algebra leads to a better intuition and a deeper understanding of the abstract concepts, and therefore simplifies their use in real world applications. Some of these applications are presented in detailed examples. In several ‘MATLAB-Minutes’ students can comprehend the concepts and results using computational experiments. Necessary basics for the use of MATLAB are presented in a short introduction. Students can also actively work with the material and practice their mathematical skills in more than 300 exerc...

  19. Linear algebra

    CERN Document Server

    Berberian, Sterling K

    2014-01-01

    Introductory treatment covers basic theory of vector spaces and linear maps - dimension, determinants, eigenvalues, and eigenvectors - plus more advanced topics such as the study of canonical forms for matrices. 1992 edition.

  20. Linear Models

    CERN Document Server

    Searle, Shayle R

    2012-01-01

    This 1971 classic on linear models is once again available--as a Wiley Classics Library Edition. It features material that can be understood by any statistician who understands matrix algebra and basic statistical methods.

  1. LINEAR ACCELERATOR

    Science.gov (United States)

    Christofilos, N.C.; Polk, I.J.

    1959-02-17

    Improvements in linear particle accelerators are described. A drift tube system for a linear ion accelerator reduces gap capacity between adjacent drift tube ends. This is accomplished by reducing the ratio of the diameter of the drift tube to the diameter of the resonant cavity. Concentration of magnetic field intensity at the longitudinal midpoint of the external sunface of each drift tube is reduced by increasing the external drift tube diameter at the longitudinal center region.

  2. 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

  3. Personalized multi-channel headphone sound reproduction based on active noise cancellation

    NARCIS (Netherlands)

    Schobben, D.W.E.; Aarts, R.M.

    2005-01-01

    A system for headphone signal processing is discussed which gives a listener the same impression as listening to a multi-channel loudspeaker set-up. It is important that this processing is optimized for each individual listener. If this is not the case, large localization errors may occur. In the

  4. A compressed sensing based method with support refinement for impulse noise cancelation in DSL

    KAUST Repository

    Quadeer, Ahmed Abdul

    2013-06-01

    This paper presents a compressed sensing based method to suppress impulse noise in digital subscriber line (DSL). The proposed algorithm exploits the sparse nature of the impulse noise and utilizes the carriers, already available in all practical DSL systems, for its estimation and cancelation. Specifically, compressed sensing is used for a coarse estimate of the impulse position, an a priori information based maximum aposteriori probability (MAP) metric for its refinement, followed by least squares (LS) or minimum mean square error (MMSE) estimation for estimating the impulse amplitudes. Simulation results show that the proposed scheme achieves higher rate as compared to other known sparse estimation algorithms in literature. The paper also demonstrates the superior performance of the proposed scheme compared to the ITU-T G992.3 standard that utilizes RS-coding for impulse noise refinement in DSL signals. © 2013 IEEE.

  5. Application of Adaptive Noise Cancellation for Anti-Vibration in Yield Monitor

    Directory of Open Access Journals (Sweden)

    Yan LI

    2014-04-01

    Full Text Available In the process of grain harvest, yield monitor system acquires real-time spatial distribution information of crop yield to provide important basis of decision-making for subsequent assignments of precision agriculture. The measurement accuracy has been seriously affected by Combine working vibration. Based on an innovative test platform of wheat combine harvester for yield monitor, well simulate the working vibration at the field situation; impact-based grain flow sensor with the structure of dual-parallel-beams as test terminals and using the NI (National Instrument data acquisition card to acquire signals; grain impacted frequency as fundamental frequency to process harmonic extraction, and for extracted signals, applied the improved LMS adaptive algorithm to interference cancellation, aim to eliminate interference cased by working vibration. The comparative experiment show that the maximum relative error less than 2 % under the proposed method and proved that the proposed algorithm in this paper is effective.

  6. 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...

  7. 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...

  8. A 380pW Dual Mode Optical Wake-up Receiver with Ambient Noise Cancellation.

    Science.gov (United States)

    Lim, Wootaek; Jang, Taekwang; Lee, Inhee; Kim, Hun-Seok; Sylvester, Dennis; Blaauw, David

    2016-06-01

    We present a sub-nW optical wake-up receiver for wireless sensor nodes. The wake-up receiver supports dual mode operation for both ultra-low standby power and high data rates, while canceling ambient in-band noise. In 0.18µm CMOS the receiver consumes 380pW in always-on wake-up mode and 28.1µW in fast RX mode at 250kbps.

  9. dc SQUID electronics based on adaptive noise cancellation and a high open-loop gain controller

    International Nuclear Information System (INIS)

    Seppae, H.

    1992-01-01

    A low-noise SQUID readout electronics with a high slew rate and an automatic gain control feature has been developed. Flux noise levels of 5x10 -7 Φ 0 /√Hz at 1 kHz and 2x10 -6 Φ 0 /√Hz at 1 Hz have been measured with this readout scheme. The system tolerates sinusoidal disturbances having amplitudes up to 140 Φ 0 at 1 kHz without loosing lock. The electronics utilizes a cooled GaAs FET to control the cancellation of the voltage noise of the room temperature amplifier, a PI 3/2 controller to provide a high open-loop gain at low frequencies, and a square-wave flux and offset voltage modulation to enable automatic control of the noise reduction. The cutoff frequency of the flux-locked-loop is 300 kHz and the feedback gain is more than 130 dB at 10 Hz. (orig.)

  10. Pump-probe differencing technique for cavity-enhanced, noise-canceling saturation laser spectroscopy.

    Science.gov (United States)

    de Vine, Glenn; McClelland, David E; Gray, Malcolm B; Close, John D

    2005-05-15

    We present an experimental technique that permits mechanical-noise-free, cavity-enhanced frequency measurements of an atomic transition and its hyperfine structure. We employ the 532-nm frequency-doubled output from a Nd:YAG laser and an iodine vapor cell. The cell is placed in a folded ring cavity (FRC) with counterpropagating pump and probe beams. The FRC is locked with the Pound-Drever-Hall technique. Mechanical noise is rejected by differencing the pump and probe signals. In addition, this differenced error signal provides a sensitive measure of differential nonlinearity within the FRC.

  11. Impulse Noise Cancellation of Medical Images Using Wavelet Networks and Median Filters

    Science.gov (United States)

    Sadri, Amir Reza; Zekri, Maryam; Sadri, Saeid; Gheissari, Niloofar

    2012-01-01

    This paper presents a new two-stage approach to impulse noise removal for medical images based on wavelet network (WN). The first step is noise detection, in which the so-called gray-level difference and average background difference are considered as the inputs of a WN. Wavelet Network is used as a preprocessing for the second stage. The second step is removing impulse noise with a median filter. The wavelet network presented here is a fixed one without learning. Experimental results show that our method acts on impulse noise effectively, and at the same time preserves chromaticity and image details very well. PMID:23493998

  12. All-mechanical quantum noise cancellation for accelerometry: broadband with momentum measurements, narrow band without

    International Nuclear Information System (INIS)

    Jacobs, Kurt; Balu, Radhakrishnan; Tezak, Nikolas; Mabuchi, Hideo

    2016-01-01

    We show that the ability to make direct measurements of momentum, in addition to the usual direct measurements of position, allows a simple configuration of two identical mechanical oscillators to be used for broadband back-action-free force metrology. This would eliminate the need for an optical reference oscillator in the scheme of Tsang and Caves (2010 Phys. Rev. Lett.  105 123601), along with its associated disadvantages. We also show that if one is restricted to position measurements alone then two copies of the same two-oscillator configuration can be used for narrow-band back-action-free force metrology. (paper)

  13. IIR digital filter design for powerline noise cancellation of ECG signal using arduino platform

    Science.gov (United States)

    Rahmatillah, Akif; Ataulkarim

    2017-05-01

    Powerline noise has been one of significant noises of Electrocardiogram (ECG) signal measurement. This noise is characterized by a sinusoidal signal which has 50 Hz of noise and 0.3 mV of maximum amplitude. This paper describes the design of IIR Notch filter design to reject a 50 Hz power line noise. IIR filter coefficients were calculated using pole placement method with three variations of band stop cut off frequencies of (49-51)Hz, (48 - 52)Hz, and (47 - 53)Hz. The algorithm and coefficients of filter were embedded to Arduino DUE (ARM 32 bit microcontroller). IIR notch filter designed has been able to reject power line noise with average square of error value of 0.225 on (49-51) Hz filter design and 0.2831 on (48 - 52)Hz filter design.

  14. Investigation of Diesel’s Residual Noise on Predictive Vehicles Noise Cancelling using LMS Adaptive Algorithm

    Science.gov (United States)

    Arttini Dwi Prasetyowati, Sri; Susanto, Adhi; Widihastuti, Ida

    2017-04-01

    Every noise problems require different solution. In this research, the noise that must be cancelled comes from roadway. Least Mean Square (LMS) adaptive is one of the algorithm that can be used to cancel that noise. Residual noise always appears and could not be erased completely. This research aims to know the characteristic of residual noise from vehicle’s noise and analysis so that it is no longer appearing as a problem. LMS algorithm was used to predict the vehicle’s noise and minimize the error. The distribution of the residual noise could be observed to determine the specificity of the residual noise. The statistic of the residual noise close to normal distribution with = 0,0435, = 1,13 and the autocorrelation of the residual noise forming impulse. As a conclusion the residual noise is insignificant.

  15. Wide-band CMOS low-noise amplifier exploiting thermal noise canceling

    NARCIS (Netherlands)

    Bruccoleri, F.; Klumperink, Eric A.M.; Nauta, Bram

    Known elementary wide-band amplifiers suffer from a fundamental tradeoff between noise figure (NF) and source impedance matching, which limits the NF to values typically above 3 dB. Global negative feedback can be used to break this tradeoff, however, at the price of potential instability. In

  16. Wideband Balun-LNA with Simultaneous Output Balancing, Noise-Canceling and Distortion-Canceling

    NARCIS (Netherlands)

    Blaakmeer, S.C.; Klumperink, Eric A.M.; Leenaerts, D.M.W.; Nauta, Bram

    2008-01-01

    An inductorless low-noise amplifier (LNA) with active balun is proposed for multi-standard radio applications between 100 MHz and 6 GHz. It exploits a combination of a common-gate (CGH) stage and an admittance-scaled common-source (CS) stage with replica biasing to maximize balanced operation, while

  17. On-line adaptive line frequency noise cancellation from a nuclear power measuring channel

    Directory of Open Access Journals (Sweden)

    Qadir Javed

    2011-01-01

    Full Text Available On-line software for adaptively canceling 50 Hz line frequency noise has been designed and tested at Pakistan Research Reactor 1. Line frequency noise causes much problem in weak signals acquisition. Sometimes this noise is so dominant that original signal is totally corrupted. Although notch filter can be used for eliminating this noise, but if signal of interest is in close vicinity of 50 Hz, then original signal is also attenuated and hence overall performance is degraded. Adaptive noise removal is a technique which could be employed for removing line frequency without degrading the desired signal. In this paper line frequency noise has been eliminated on-line from a nuclear power measuring channel. The adaptive LMS algorithm has been used to cancel 50 Hz noise. The algorithm has been implemented in labVIEW with NI 6024 data acquisition card. The quality of the acquired signal has been improved much as can be seen in experimental results.

  18. A wideband CMOS inductorless low noise amplifier employing noise cancellation for digital TV tuner applications

    International Nuclear Information System (INIS)

    Zhang Jihong; Bai Xuefei; Huang Lu

    2013-01-01

    A wideband inductorless low noise amplifier for digital TV tuner applications is presented. The proposed LNA scheme uses a composite NMOS/PMOS cross-coupled transistor pair to provide partial cancellation of noise generated by the input transistors. The chip is implemented in SMIC 0.18 μm CMOS technology. Measurement shows that the proposed LNA achieves 12.2–15.2 dB voltage gain from 300 to 900 MHz, the noise figure is below 3.1 dB and has a minimum value of 2.3 dB, and the best input-referred 1-dB compression point (IP1dB) is − 17 dBm at 900 MHz. The core consumes 7 mA current with a supply voltage of 1.8 V and occupies an area of 0.5 × 0.35 mm 2 . (semiconductor integrated circuits)

  19. An LCMV Filter for Single-Channel Noise Cancellation and Reduction in the Time Domain

    DEFF Research Database (Denmark)

    Jensen, Jesper Rindom; Benesty, Jacob; Christensen, Mads Græsbøll

    2013-01-01

    In this paper, we consider a recent class of optimal rectangular fil- tering matrices for single-channel speech enhancement. This class of filters exploits the fact that the dimension of the signal subspace is lower than that of the full space. Then, extra degrees of freedom in the filters...... signal-to-interference ratio. This is showed for both synthetic and real speech signals....

  20. Linear regression

    CERN Document Server

    Olive, David J

    2017-01-01

    This text covers both multiple linear regression and some experimental design models. The text uses the response plot to visualize the model and to detect outliers, does not assume that the error distribution has a known parametric distribution, develops prediction intervals that work when the error distribution is unknown, suggests bootstrap hypothesis tests that may be useful for inference after variable selection, and develops prediction regions and large sample theory for the multivariate linear regression model that has m response variables. A relationship between multivariate prediction regions and confidence regions provides a simple way to bootstrap confidence regions. These confidence regions often provide a practical method for testing hypotheses. There is also a chapter on generalized linear models and generalized additive models. There are many R functions to produce response and residual plots, to simulate prediction intervals and hypothesis tests, to detect outliers, and to choose response trans...

  1. Linear Colliders

    International Nuclear Information System (INIS)

    Alcaraz, J.

    2001-01-01

    After several years of study e''+ e''- linear colliders in the TeV range have emerged as the major and optimal high-energy physics projects for the post-LHC era. These notes summarize the present status form the main accelerator and detector features to their physics potential. The LHC era. These notes summarize the present status, from the main accelerator and detector features to their physics potential. The LHC is expected to provide first discoveries in the new energy domain, whereas an e''+ e''- linear collider in the 500 GeV-1 TeV will be able to complement it to an unprecedented level of precision in any possible areas: Higgs, signals beyond the SM and electroweak measurements. It is evident that the Linear Collider program will constitute a major step in the understanding of the nature of the new physics beyond the Standard Model. (Author) 22 refs

  2. Linear algebra

    CERN Document Server

    Edwards, Harold M

    1995-01-01

    In his new undergraduate textbook, Harold M Edwards proposes a radically new and thoroughly algorithmic approach to linear algebra Originally inspired by the constructive philosophy of mathematics championed in the 19th century by Leopold Kronecker, the approach is well suited to students in the computer-dominated late 20th century Each proof is an algorithm described in English that can be translated into the computer language the class is using and put to work solving problems and generating new examples, making the study of linear algebra a truly interactive experience Designed for a one-semester course, this text adopts an algorithmic approach to linear algebra giving the student many examples to work through and copious exercises to test their skills and extend their knowledge of the subject Students at all levels will find much interactive instruction in this text while teachers will find stimulating examples and methods of approach to the subject

  3. 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)

  4. 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

  5. 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)

  6. 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.

  7. Interface for Barge-in Free Spoken Dialogue System Based on Sound Field Reproduction and Microphone Array

    Directory of Open Access Journals (Sweden)

    Hinamoto Yoichi

    2007-01-01

    Full Text Available A barge-in free spoken dialogue interface using sound field control and microphone array is proposed. In the conventional spoken dialogue system using an acoustic echo canceller, it is indispensable to estimate a room transfer function, especially when the transfer function is changed by various interferences. However, the estimation is difficult when the user and the system speak simultaneously. To resolve the problem, we propose a sound field control technique to prevent the response sound from being observed. Combined with a microphone array, the proposed method can achieve high elimination performance with no adaptive process. The efficacy of the proposed interface is ascertained in the experiments on the basis of sound elimination and speech recognition.

  8. 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.

  9. 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.

  10. 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.

  11. 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.

  12. 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...

  13. 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.

  14. 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.

  15. 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...

  16. 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

  17. 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.

  18. 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.

  19. Detection System of Sound Noise Level (SNL) Based on Condenser Microphone Sensor

    Science.gov (United States)

    Rajagukguk, Juniastel; Eka Sari, Nurdieni

    2018-03-01

    The research aims to know the noise level by using the Arduino Uno as data processing input from sensors and called as Sound Noise Level (SNL). The working principle of the instrument is as noise detector with the show notifications the noise level on the LCD indicator and in the audiovisual form. Noise detection using the sensor is a condenser microphone and LM 567 as IC op-amps, which are assembled so that it can detect the noise, which sounds are captured by the sensor will turn the tide of sinusoida voice became sine wave energy electricity (altering sinusoida electric current) that is able to responded to complaints by the Arduino Uno. The tool is equipped with a detector consists of a set indicator LED and sound well as the notification from the text on LCD 16*2. Work setting indicators on the condition that, if the measured noise > 75 dB then sound will beep, the red LED will light up indicating the status of the danger. If the measured value on the LCD is higher than 56 dB, sound indicator will be beep and yellow LED will be on indicating noisy. If the noise measured value <55 dB, sound indicator will be quiet indicating peaceful from noisy. From the result of the research can be explained that the SNL is capable to detecting and displaying noise level with a measuring range 50-100 dB and capable to delivering the notification noise in audiovisual.

  20. 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.

  1. Linear programming

    CERN Document Server

    Karloff, Howard

    1991-01-01

    To this reviewer’s knowledge, this is the first book accessible to the upper division undergraduate or beginning graduate student that surveys linear programming from the Simplex Method…via the Ellipsoid algorithm to Karmarkar’s algorithm. Moreover, its point of view is algorithmic and thus it provides both a history and a case history of work in complexity theory. The presentation is admirable; Karloff's style is informal (even humorous at times) without sacrificing anything necessary for understanding. Diagrams (including horizontal brackets that group terms) aid in providing clarity. The end-of-chapter notes are helpful...Recommended highly for acquisition, since it is not only a textbook, but can also be used for independent reading and study. —Choice Reviews The reader will be well served by reading the monograph from cover to cover. The author succeeds in providing a concise, readable, understandable introduction to modern linear programming. —Mathematics of Computing This is a textbook intend...

  2. Reduction of Linear Programming to Linear Approximation

    OpenAIRE

    Vaserstein, Leonid N.

    2006-01-01

    It is well known that every Chebyshev linear approximation problem can be reduced to a linear program. In this paper we show that conversely every linear program can be reduced to a Chebyshev linear approximation problem.

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

    Science.gov (United States)

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

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

  4. Automatic detection of whole night snoring events using non-contact microphone.

    Directory of Open Access Journals (Sweden)

    Eliran Dafna

    Full Text Available OBJECTIVE: Although awareness of sleep disorders is increasing, limited information is available on whole night detection of snoring. Our study aimed to develop and validate a robust, high performance, and sensitive whole-night snore detector based on non-contact technology. DESIGN: Sounds during polysomnography (PSG were recorded using a directional condenser microphone placed 1 m above the bed. An AdaBoost classifier was trained and validated on manually labeled snoring and non-snoring acoustic events. PATIENTS: Sixty-seven subjects (age 52.5 ± 13.5 years, BMI 30.8 ± 4.7 kg/m(2, m/f 40/27 referred for PSG for obstructive sleep apnea diagnoses were prospectively and consecutively recruited. Twenty-five subjects were used for the design study; the validation study was blindly performed on the remaining forty-two subjects. MEASUREMENTS AND RESULTS: To train the proposed sound detector, >76,600 acoustic episodes collected in the design study were manually classified by three scorers into snore and non-snore episodes (e.g., bedding noise, coughing, environmental. A feature selection process was applied to select the most discriminative features extracted from time and spectral domains. The average snore/non-snore detection rate (accuracy for the design group was 98.4% based on a ten-fold cross-validation technique. When tested on the validation group, the average detection rate was 98.2% with sensitivity of 98.0% (snore as a snore and specificity of 98.3% (noise as noise. CONCLUSIONS: Audio-based features extracted from time and spectral domains can accurately discriminate between snore and non-snore acoustic events. This audio analysis approach enables detection and analysis of snoring sounds from a full night in order to produce quantified measures for objective follow-up of patients.

  5. linear-quadratic-linear model

    Directory of Open Access Journals (Sweden)

    Tanwiwat Jaikuna

    2017-02-01

    Full Text Available Purpose: To develop an in-house software program that is able to calculate and generate the biological dose distribution and biological dose volume histogram by physical dose conversion using the linear-quadratic-linear (LQL model. Material and methods : The Isobio software was developed using MATLAB version 2014b to calculate and generate the biological dose distribution and biological dose volume histograms. The physical dose from each voxel in treatment planning was extracted through Computational Environment for Radiotherapy Research (CERR, and the accuracy was verified by the differentiation between the dose volume histogram from CERR and the treatment planning system. An equivalent dose in 2 Gy fraction (EQD2 was calculated using biological effective dose (BED based on the LQL model. The software calculation and the manual calculation were compared for EQD2 verification with pair t-test statistical analysis using IBM SPSS Statistics version 22 (64-bit. Results: Two and three-dimensional biological dose distribution and biological dose volume histogram were displayed correctly by the Isobio software. Different physical doses were found between CERR and treatment planning system (TPS in Oncentra, with 3.33% in high-risk clinical target volume (HR-CTV determined by D90%, 0.56% in the bladder, 1.74% in the rectum when determined by D2cc, and less than 1% in Pinnacle. The difference in the EQD2 between the software calculation and the manual calculation was not significantly different with 0.00% at p-values 0.820, 0.095, and 0.593 for external beam radiation therapy (EBRT and 0.240, 0.320, and 0.849 for brachytherapy (BT in HR-CTV, bladder, and rectum, respectively. Conclusions : The Isobio software is a feasible tool to generate the biological dose distribution and biological dose volume histogram for treatment plan evaluation in both EBRT and BT.

  6. 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.

  7. 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....

  8. 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...

  9. 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

  10. 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.

  11. 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.

  12. 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.

  13. 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.

  14. Linear Algebra and Smarandache Linear Algebra

    OpenAIRE

    Vasantha, Kandasamy

    2003-01-01

    The present book, on Smarandache linear algebra, not only studies the Smarandache analogues of linear algebra and its applications, it also aims to bridge the need for new research topics pertaining to linear algebra, purely in the algebraic sense. We have introduced Smarandache semilinear algebra, Smarandache bilinear algebra and Smarandache anti-linear algebra and their fuzzy equivalents. Moreover, in this book, we have brought out the study of linear algebra and vector spaces over finite p...

  15. 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...

  16. 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...

  17. Linear Regression on Sparse Features for Single-Channel Speech Separation

    DEFF Research Database (Denmark)

    Schmidt, Mikkel N.; Olsson, Rasmus Kongsgaard

    2007-01-01

    In this work we address the problem of separating multiple speakers from a single microphone recording. We formulate a linear regression model for estimating each speaker based on features derived from the mixture. The employed feature representation is a sparse, non-negative encoding of the speech...... mixture in terms of pre-learned speaker-dependent dictionaries. Previous work has shown that this feature representation by itself provides some degree of separation. We show that the performance is significantly improved when regression analysis is performed on the sparse, non-negative features, both...

  18. 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.

  19. 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?

  20. 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.

  1. 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

  2. Linearly constrained minimax optimization

    DEFF Research Database (Denmark)

    Madsen, Kaj; Schjær-Jacobsen, Hans

    1978-01-01

    We present an algorithm for nonlinear minimax optimization subject to linear equality and inequality constraints which requires first order partial derivatives. The algorithm is based on successive linear approximations to the functions defining the problem. The resulting linear subproblems...

  3. Foundations of linear and generalized linear models

    CERN Document Server

    Agresti, Alan

    2015-01-01

    A valuable overview of the most important ideas and results in statistical analysis Written by a highly-experienced author, Foundations of Linear and Generalized Linear Models is a clear and comprehensive guide to the key concepts and results of linear statistical models. The book presents a broad, in-depth overview of the most commonly used statistical models by discussing the theory underlying the models, R software applications, and examples with crafted models to elucidate key ideas and promote practical model building. The book begins by illustrating the fundamentals of linear models,

  4. 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

  5. Algebraically approximate and noisy realization of discrete-time systems and digital images

    CERN Document Server

    Hasegawa, Yasumichi

    2009-01-01

    This monograph deals with approximation and noise cancellation of dynamical systems which include linear and nonlinear input/output relationships. It also deal with approximation and noise cancellation of two dimensional arrays. It will be of special interest to researchers, engineers and graduate students who have specialized in filtering theory and system theory and digital images. This monograph is composed of two parts. Part I and Part II will deal with approximation and noise cancellation of dynamical systems or digital images respectively. From noiseless or noisy data, reduction will be

  6. 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....

  7. Validation study of two-microphone acoustic reflectometry for determination of breathing tube placement in 200 adult patients.

    Science.gov (United States)

    Raphael, David T; Benbassat, Maxim; Arnaudov, Dimiter; Bohorquez, Alex; Nasseri, Bita

    2002-12-01

    Acoustic reflectometry allows the construction of a one-dimensional image of a cavity, such as the airway or the esophagus. The reflectometric area-distance profile consists of a constant cross-sectional area segment (length of endotracheal tube), followed either by a rapid increase in the area beyond the carina (tracheal intubation) or by an immediate decrease in the area (esophageal intubation). Two hundred adult patients were induced and intubated, without restrictions on anesthetic agents or airway adjunct devices. A two-microphone acoustic reflectometer was used to determine whether the breathing tube was placed in the trachea or esophagus. A blinded reflectometer operator, seated a distance away from the patient, interpreted the acoustic area-distance profile alone to decide where the tube was placed. Capnography was used as the gold standard. Of 200 tracheal intubations confirmed by capnography, the reflectometer operator correctly identified 198 (99% correct tracheal intubation identification rate). In two patients there were false-negative results, patients with a tracheal intubation were interpreted as having an esophageal intubation. A total of 14 esophageal intubations resulted, all correctly identified by reflectometry, for a 100% esophageal intubation identification rate. Acoustic reflectometry is a rapid, noninvasive method by which to determine whether breathing tube placement is correct (tracheal) or incorrect (esophageal). Reflectometry determination of tube placement may be useful in airway emergencies, particularly in cases where visualization of the glottic area is not possible and capnography may fail, as in patients with cardiac arrest.

  8. Frequency-Dependent Amplitude Panning for the Stereophonic Image Enhancement of Audio Recorded Using Two Closely Spaced Microphones

    Directory of Open Access Journals (Sweden)

    Chan Jun Chun

    2016-02-01

    Full Text Available In this paper, we propose a new frequency-dependent amplitude panning method for stereophonic image enhancement applied to a sound source recorded using two closely spaced omni-directional microphones. The ability to detect the direction of such a sound source is limited due to weak spatial information, such as the inter-channel time difference (ICTD and inter-channel level difference (ICLD. Moreover, when sound sources are recorded in a convolutive or a real room environment, the detection of sources is affected by reverberation effects. Thus, the proposed method first tries to estimate the source direction depending on the frequency using azimuth-frequency analysis. Then, a frequency-dependent amplitude panning technique is proposed to enhance the stereophonic image by modifying the stereophonic law of sines. To demonstrate the effectiveness of the proposed method, we compare its performance with that of a conventional method based on the beamforming technique in terms of directivity pattern, perceived direction, and quality degradation under three different recording conditions (anechoic, convolutive, and real reverberant. The comparison shows that the proposed method gives us better stereophonic images in a stereo loudspeaker reproduction than the conventional method without any annoying effects.

  9. 3D-Printing of inverted pyramid suspending architecture for pyroelectric infrared detectors with inhibited microphonic effect

    Science.gov (United States)

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

    2016-05-01

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

  10. 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.

  11. 2015 AMCLC Open-Microphone Session: Improving the IQ (Information Quality) of What We Do.

    Science.gov (United States)

    Stern, Eric; Metter, Darlene; Everett, Catherine; Flug, Jonathan; Herrington, William; Applegate, Kimberly E

    2016-03-01

    Each year an open-microphone session is hosted by the Council Steering Committee. The committee invited an expert panel to discuss the use of effective communication in appropriate procedure selection and methods and resources to communicate the results of procedures performed in an actionable and clear manner to referring clinicians and patients, as well as downstream data systems. The ACR is actively encouraging radiologists to leverage existing and new technologies to increase their visibility in the health care system. Key features in Imaging 3.0 are results reporting through actionable reports, decision support for results reporting, guidelines for recommendations, tools for actionable reports, and tracking a radiologist's recommendations. The final radiology report is an essential product of our service, but it is increasingly clear that the noninterpretive components of our profession will add the most value to patient care. The radiology report is not the only evidence of our work. Nonetheless, the information quality and content of the radiology report can and must be improved so that it can add value and clinical usefulness toward excellent patient care. We must use appropriate tools and "best knowledge" to deliver actionable and value-added high-quality reports. Copyright © 2016 American College of Radiology. Published by Elsevier Inc. All rights reserved.

  12. Passive synthetic aperture sonar techniques in combination with tow ship noise canceling: application to a triplet towed array

    NARCIS (Netherlands)

    Colin, M.E.G.D.; Groen, J.

    2002-01-01

    An important issue in research on passive ASW operations is improvement in signal-to-noise ratio (SNR) and bearing resolution for targets emitting low frequency signals. One of the techniques believed to improve these characteristics is Synthetic Aperture Sonar (SAS). The method is based on the

  13. 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

  14. A linear programming manual

    Science.gov (United States)

    Tuey, R. C.

    1972-01-01

    Computer solutions of linear programming problems are outlined. Information covers vector spaces, convex sets, and matrix algebra elements for solving simultaneous linear equations. Dual problems, reduced cost analysis, ranges, and error analysis are illustrated.

  15. Linear shaped charge

    Energy Technology Data Exchange (ETDEWEB)

    Peterson, David; Stofleth, Jerome H.; Saul, Venner W.

    2017-07-11

    Linear shaped charges are described herein. In a general embodiment, the linear shaped charge has an explosive with an elongated arrowhead-shaped profile. The linear shaped charge also has and an elongated v-shaped liner that is inset into a recess of the explosive. Another linear shaped charge includes an explosive that is shaped as a star-shaped prism. Liners are inset into crevices of the explosive, where the explosive acts as a tamper.

  16. Classifying Linear Canonical Relations

    OpenAIRE

    Lorand, Jonathan

    2015-01-01

    In this Master's thesis, we consider the problem of classifying, up to conjugation by linear symplectomorphisms, linear canonical relations (lagrangian correspondences) from a finite-dimensional symplectic vector space to itself. We give an elementary introduction to the theory of linear canonical relations and present partial results toward the classification problem. This exposition should be accessible to undergraduate students with a basic familiarity with linear algebra.

  17. Linear-Algebra Programs

    Science.gov (United States)

    Lawson, C. L.; Krogh, F. T.; Gold, S. S.; Kincaid, D. R.; Sullivan, J.; Williams, E.; Hanson, R. J.; Haskell, K.; Dongarra, J.; Moler, C. B.

    1982-01-01

    The Basic Linear Algebra Subprograms (BLAS) library is a collection of 38 FORTRAN-callable routines for performing basic operations of numerical linear algebra. BLAS library is portable and efficient source of basic operations for designers of programs involving linear algebriac computations. BLAS library is supplied in portable FORTRAN and Assembler code versions for IBM 370, UNIVAC 1100 and CDC 6000 series computers.

  18. Non linear system become linear system

    Directory of Open Access Journals (Sweden)

    Petre Bucur

    2007-01-01

    Full Text Available The present paper refers to the theory and the practice of the systems regarding non-linear systems and their applications. We aimed the integration of these systems to elaborate their response as well as to highlight some outstanding features.

  19. Linear motor coil assembly and linear motor

    NARCIS (Netherlands)

    2009-01-01

    An ironless linear motor (5) comprising a magnet track (53) and a coil assembly (50) operating in cooperation with said magnet track (53) and having a plurality of concentrated multi-turn coils (31 a-f, 41 a-d, 51 a-k), wherein the end windings (31E) of the coils (31 a-f, 41 a-e) are substantially

  20. Linear collider: a preview

    Energy Technology Data Exchange (ETDEWEB)

    Wiedemann, H.

    1981-11-01

    Since no linear colliders have been built yet it is difficult to know at what energy the linear cost scaling of linear colliders drops below the quadratic scaling of storage rings. There is, however, no doubt that a linear collider facility for a center of mass energy above say 500 GeV is significantly cheaper than an equivalent storage ring. In order to make the linear collider principle feasible at very high energies a number of problems have to be solved. There are two kinds of problems: one which is related to the feasibility of the principle and the other kind of problems is associated with minimizing the cost of constructing and operating such a facility. This lecture series describes the problems and possible solutions. Since the real test of a principle requires the construction of a prototype I will in the last chapter describe the SLC project at the Stanford Linear Accelerator Center.

  1. Basic linear algebra

    CERN Document Server

    Blyth, T S

    2002-01-01

    Basic Linear Algebra is a text for first year students leading from concrete examples to abstract theorems, via tutorial-type exercises. More exercises (of the kind a student may expect in examination papers) are grouped at the end of each section. The book covers the most important basics of any first course on linear algebra, explaining the algebra of matrices with applications to analytic geometry, systems of linear equations, difference equations and complex numbers. Linear equations are treated via Hermite normal forms which provides a successful and concrete explanation of the notion of linear independence. Another important highlight is the connection between linear mappings and matrices leading to the change of basis theorem which opens the door to the notion of similarity. This new and revised edition features additional exercises and coverage of Cramer's rule (omitted from the first edition). However, it is the new, extra chapter on computer assistance that will be of particular interest to readers:...

  2. Linear collider: a preview

    International Nuclear Information System (INIS)

    Wiedemann, H.

    1981-11-01

    Since no linear colliders have been built yet it is difficult to know at what energy the linear cost scaling of linear colliders drops below the quadratic scaling of storage rings. There is, however, no doubt that a linear collider facility for a center of mass energy above say 500 GeV is significantly cheaper than an equivalent storage ring. In order to make the linear collider principle feasible at very high energies a number of problems have to be solved. There are two kinds of problems: one which is related to the feasibility of the principle and the other kind of problems is associated with minimizing the cost of constructing and operating such a facility. This lecture series describes the problems and possible solutions. Since the real test of a principle requires the construction of a prototype I will in the last chapter describe the SLC project at the Stanford Linear Accelerator Center

  3. 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.

  4. Matrices and linear transformations

    CERN Document Server

    Cullen, Charles G

    1990-01-01

    ""Comprehensive . . . an excellent introduction to the subject."" - Electronic Engineer's Design Magazine.This introductory textbook, aimed at sophomore- and junior-level undergraduates in mathematics, engineering, and the physical sciences, offers a smooth, in-depth treatment of linear algebra and matrix theory. The major objects of study are matrices over an arbitrary field. Contents include Matrices and Linear Systems; Vector Spaces; Determinants; Linear Transformations; Similarity: Part I and Part II; Polynomials and Polynomial Matrices; Matrix Analysis; and Numerical Methods. The first

  5. Efficient Non Linear Loudspeakers

    DEFF Research Database (Denmark)

    Petersen, Bo R.; Agerkvist, Finn T.

    2006-01-01

    Loudspeakers have traditionally been designed to be as linear as possible. However, as techniques for compensating non linearities are emerging, it becomes possible to use other design criteria. This paper present and examines a new idea for improving the efficiency of loudspeakers at high levels...... by changing the voice coil layout. This deliberate non-linear design has the benefit that a smaller amplifier can be used, which has the benefit of reducing system cost as well as reducing power consumption....

  6. Linear models with R

    CERN Document Server

    Faraway, Julian J

    2014-01-01

    A Hands-On Way to Learning Data AnalysisPart of the core of statistics, linear models are used to make predictions and explain the relationship between the response and the predictors. Understanding linear models is crucial to a broader competence in the practice of statistics. Linear Models with R, Second Edition explains how to use linear models in physical science, engineering, social science, and business applications. The book incorporates several improvements that reflect how the world of R has greatly expanded since the publication of the first edition.New to the Second EditionReorganiz

  7. Linear integrated circuits

    CERN Document Server

    Carr, Joseph

    1996-01-01

    The linear IC market is large and growing, as is the demand for well trained technicians and engineers who understand how these devices work and how to apply them. Linear Integrated Circuits provides in-depth coverage of the devices and their operation, but not at the expense of practical applications in which linear devices figure prominently. This book is written for a wide readership from FE and first degree students, to hobbyists and professionals.Chapter 1 offers a general introduction that will provide students with the foundations of linear IC technology. From chapter 2 onwa

  8. Fault tolerant linear actuator

    Science.gov (United States)

    Tesar, Delbert

    2004-09-14

    In varying embodiments, the fault tolerant linear actuator of the present invention is a new and improved linear actuator with fault tolerance and positional control that may incorporate velocity summing, force summing, or a combination of the two. In one embodiment, the invention offers a velocity summing arrangement with a differential gear between two prime movers driving a cage, which then drives a linear spindle screw transmission. Other embodiments feature two prime movers driving separate linear spindle screw transmissions, one internal and one external, in a totally concentric and compact integrated module.

  9. Superconducting linear accelerator cryostat

    International Nuclear Information System (INIS)

    Ben-Zvi, I.; Elkonin, B.V.; Sokolowski, J.S.

    1984-01-01

    A large vertical cryostat for a superconducting linear accelerator using quarter wave resonators has been developed. The essential technical details, operational experience and performance are described. (author)

  10. Linearity enigmas in ecology

    Energy Technology Data Exchange (ETDEWEB)

    Patten, B.C.

    1983-04-01

    Two issues concerning linearity or nonlinearity of natural systems are considered. Each is related to one of the two alternative defining properties of linear systems, superposition and decomposition. Superposition exists when a linear combination of inputs to a system results in the same linear combination of outputs that individually correspond to the original inputs. To demonstrate this property it is necessary that all initial states and inputs of the system which impinge on the output in question be included in the linear combination manipulation. As this is difficult or impossible to do with real systems of any complexity, nature appears nonlinear even though it may be linear. A linear system that displays nonlinear behavior for this reason is termed pseudononlinear. The decomposition property exists when the dynamic response of a system can be partitioned into an input-free portion due to state plus a state-free portion due to input. This is a characteristic of all linear systems, but not of nonlinear systems. Without the decomposition property, it is not possible to distinguish which portions of a system's behavior are due to innate characteristics (self) vs. outside conditions (environment), which is an important class of questions in biology and ecology. Some philosophical aspects of these findings are then considered. It is suggested that those ecologists who hold to the view that organisms and their environments are separate entities are in effect embracing a linear view of nature, even though their belief systems and mathematical models tend to be nonlinear. On the other hand, those who consider that organism-environment complex forms a single inseparable unit are implictly involved in non-linear thought, which may be in conflict with the linear modes and models that some of them use. The need to rectify these ambivalences on the part of both groups is indicated.

  11. A critical review of hearing-aid single-microphone noise-reduction studies in adults and children.

    Science.gov (United States)

    Chong, Foong Yen; Jenstad, Lorienne M

    2017-10-26

    Single-microphone noise reduction (SMNR) is implemented in hearing aids to suppress background noise. The purpose of this article was to provide a critical review of peer-reviewed studies in adults and children with sensorineural hearing loss who were fitted with hearing aids incorporating SMNR. Articles published between 2000 and 2016 were searched in PUBMED and EBSCO databases. Thirty-two articles were included in the final review. Most studies with adult participants showed that SMNR has no effect on speech intelligibility. Positive results were reported for acceptance of background noise, preference, and listening effort. Studies of school-aged children were consistent with the findings of adult studies. No study with infants or young children of under 5 years old was found. Recent studies on noise-reduction systems not yet available in wearable hearing aids have documented benefits of noise reduction on memory for speech processing for older adults. This evidence supports the use of SMNR for adults and school-aged children when the aim is to improve listening comfort or reduce listening effort. Future research should test SMNR with infants and children who are younger than 5 years of age. Further development, testing, and clinical trials should be carried out on algorithms not yet available in wearable hearing aids. Testing higher cognitive level for speech processing and learning of novel sounds or words could show benefits of advanced signal processing features. These approaches should be expanded to other populations such as children and younger adults. Implications for rehabilitation The review provides a quick reference for students and clinicians regarding the efficacy and effectiveness of SMNR in wearable hearing aids. This information is useful during counseling session to build a realistic expectation among hearing aid users. Most studies in the adult population suggest that SMNR may provide some benefits to adult listeners in terms of listening

  12. Linear colliders - prospects 1985

    International Nuclear Information System (INIS)

    Rees, J.

    1985-06-01

    We discuss the scaling laws of linear colliders and their consequences for accelerator design. We then report on the SLAC Linear Collider project and comment on experience gained on that project and its application to future colliders. 9 refs., 2 figs

  13. The SLAC linear collider

    International Nuclear Information System (INIS)

    Richter, B.

    1985-01-01

    A report is given on the goals and progress of the SLAC Linear Collider. The author discusses the status of the machine and the detectors and give an overview of the physics which can be done at this new facility. He also gives some ideas on how (and why) large linear colliders of the future should be built

  14. Linear Programming (LP)

    International Nuclear Information System (INIS)

    Rogner, H.H.

    1989-01-01

    The submitted sections on linear programming are extracted from 'Theorie und Technik der Planung' (1978) by W. Blaas and P. Henseler and reformulated for presentation at the Workshop. They consider a brief introduction to the theory of linear programming and to some essential aspects of the SIMPLEX solution algorithm for the purposes of economic planning processes. 1 fig

  15. Racetrack linear accelerators

    International Nuclear Information System (INIS)

    Rowe, C.H.; Wilton, M.S. de.

    1979-01-01

    An improved recirculating electron beam linear accelerator of the racetrack type is described. The system comprises a beam path of four straight legs with four Pretzel bending magnets at the end of each leg to direct the beam into the next leg of the beam path. At least one of the beam path legs includes a linear accelerator. (UK)

  16. Semidefinite linear complementarity problems

    International Nuclear Information System (INIS)

    Eckhardt, U.

    1978-04-01

    Semidefinite linear complementarity problems arise by discretization of variational inequalities describing e.g. elastic contact problems, free boundary value problems etc. In the present paper linear complementarity problems are introduced and the theory as well as the numerical treatment of them are described. In the special case of semidefinite linear complementarity problems a numerical method is presented which combines the advantages of elimination and iteration methods without suffering from their drawbacks. This new method has very attractive properties since it has a high degree of invariance with respect to the representation of the set of all feasible solutions of a linear complementarity problem by linear inequalities. By means of some practical applications the properties of the new method are demonstrated. (orig.) [de

  17. Linear algebra done right

    CERN Document Server

    Axler, Sheldon

    2015-01-01

    This best-selling textbook for a second course in linear algebra is aimed at undergrad math majors and graduate students. The novel approach taken here banishes determinants to the end of the book. The text focuses on the central goal of linear algebra: understanding the structure of linear operators on finite-dimensional vector spaces. The author has taken unusual care to motivate concepts and to simplify proofs. A variety of interesting exercises in each chapter helps students understand and manipulate the objects of linear algebra. The third edition contains major improvements and revisions throughout the book. More than 300 new exercises have been added since the previous edition. Many new examples have been added to illustrate the key ideas of linear algebra. New topics covered in the book include product spaces, quotient spaces, and dual spaces. Beautiful new formatting creates pages with an unusually pleasant appearance in both print and electronic versions. No prerequisites are assumed other than the ...

  18. Handbook on linear motor application

    International Nuclear Information System (INIS)

    1988-10-01

    This book guides the application for Linear motor. It lists classification and speciality of Linear Motor, terms of linear-induction motor, principle of the Motor, types on one-side linear-induction motor, bilateral linear-induction motor, linear-DC Motor on basic of the motor, linear-DC Motor for moving-coil type, linear-DC motor for permanent-magnet moving type, linear-DC motor for electricity non-utility type, linear-pulse motor for variable motor, linear-pulse motor for permanent magneto type, linear-vibration actuator, linear-vibration actuator for moving-coil type, linear synchronous motor, linear electromagnetic motor, linear electromagnetic solenoid, technical organization and magnetic levitation and linear motor and sensor.

  19. Linear ubiquitination in immunity.

    Science.gov (United States)

    Shimizu, Yutaka; Taraborrelli, Lucia; Walczak, Henning

    2015-07-01

    Linear ubiquitination is a post-translational protein modification recently discovered to be crucial for innate and adaptive immune signaling. The function of linear ubiquitin chains is regulated at multiple levels: generation, recognition, and removal. These chains are generated by the linear ubiquitin chain assembly complex (LUBAC), the only known ubiquitin E3 capable of forming the linear ubiquitin linkage de novo. LUBAC is not only relevant for activation of nuclear factor-κB (NF-κB) and mitogen-activated protein kinases (MAPKs) in various signaling pathways, but importantly, it also regulates cell death downstream of immune receptors capable of inducing this response. Recognition of the linear ubiquitin linkage is specifically mediated by certain ubiquitin receptors, which is crucial for translation into the intended signaling outputs. LUBAC deficiency results in attenuated gene activation and increased cell death, causing pathologic conditions in both, mice, and humans. Removal of ubiquitin chains is mediated by deubiquitinases (DUBs). Two of them, OTULIN and CYLD, are constitutively associated with LUBAC. Here, we review the current knowledge on linear ubiquitination in immune signaling pathways and the biochemical mechanisms as to how linear polyubiquitin exerts its functions distinctly from those of other ubiquitin linkage types. © 2015 The Authors. Immunological Reviews Published by John Wiley & Sons Ltd.

  20. Linearizing W-algebras

    International Nuclear Information System (INIS)

    Krivonos, S.O.; Sorin, A.S.

    1994-06-01

    We show that the Zamolodchikov's and Polyakov-Bershadsky nonlinear algebras W 3 and W (2) 3 can be embedded as subalgebras into some linear algebras with finite set of currents. Using these linear algebras we find new field realizations of W (2) 3 and W 3 which could be a starting point for constructing new versions of W-string theories. We also reveal a number of hidden relationships between W 3 and W (2) 3 . We conjecture that similar linear algebras can exist for other W-algebra as well. (author). 10 refs

  1. Matrices and linear algebra

    CERN Document Server

    Schneider, Hans

    1989-01-01

    Linear algebra is one of the central disciplines in mathematics. A student of pure mathematics must know linear algebra if he is to continue with modern algebra or functional analysis. Much of the mathematics now taught to engineers and physicists requires it.This well-known and highly regarded text makes the subject accessible to undergraduates with little mathematical experience. Written mainly for students in physics, engineering, economics, and other fields outside mathematics, the book gives the theory of matrices and applications to systems of linear equations, as well as many related t

  2. Linearity in Process Languages

    DEFF Research Database (Denmark)

    Nygaard, Mikkel; Winskel, Glynn

    2002-01-01

    The meaning and mathematical consequences of linearity (managing without a presumed ability to copy) are studied for a path-based model of processes which is also a model of affine-linear logic. This connection yields an affine-linear language for processes, automatically respecting open......-map bisimulation, in which a range of process operations can be expressed. An operational semantics is provided for the tensor fragment of the language. Different ways to make assemblies of processes lead to different choices of exponential, some of which respect bisimulation....

  3. Elements of linear space

    CERN Document Server

    Amir-Moez, A R; Sneddon, I N

    1962-01-01

    Elements of Linear Space is a detailed treatment of the elements of linear spaces, including real spaces with no more than three dimensions and complex n-dimensional spaces. The geometry of conic sections and quadric surfaces is considered, along with algebraic structures, especially vector spaces and transformations. Problems drawn from various branches of geometry are given.Comprised of 12 chapters, this volume begins with an introduction to real Euclidean space, followed by a discussion on linear transformations and matrices. The addition and multiplication of transformations and matrices a

  4. Applied linear regression

    CERN Document Server

    Weisberg, Sanford

    2013-01-01

    Praise for the Third Edition ""...this is an excellent book which could easily be used as a course text...""-International Statistical Institute The Fourth Edition of Applied Linear Regression provides a thorough update of the basic theory and methodology of linear regression modeling. Demonstrating the practical applications of linear regression analysis techniques, the Fourth Edition uses interesting, real-world exercises and examples. Stressing central concepts such as model building, understanding parameters, assessing fit and reliability, and drawing conclusions, the new edition illus

  5. Linear system theory

    Science.gov (United States)

    Callier, Frank M.; Desoer, Charles A.

    1991-01-01

    The aim of this book is to provide a systematic and rigorous access to the main topics of linear state-space system theory in both the continuous-time case and the discrete-time case; and the I/O description of linear systems. The main thrusts of the work are the analysis of system descriptions and derivations of their properties, LQ-optimal control, state feedback and state estimation, and MIMO unity-feedback systems.

  6. Digital Low-Level RF Controls for Future Superconducting Linear Colliders

    CERN Document Server

    Simrock, Stefan

    2005-01-01

    The requirements for RF Control Systems of Superconducting Linear Colliders are not only defined in terms of the quality of field control but also with respect to operability, availability, and maintainability of the RF System, and the interfaces to other subsystems. The field control of the vector-sum of many cavities driven by one klystron in pulsed mode at high gradients is a challenging task since severe Lorentz force detuning, microphonics and beam induced field errors must be suppressed by several orders of magnitude. This is accomplished by a combination of local and global feedback and feedforward control. Sensors monitor individual cavity probe signals, and forward and reflected wave as well as the beam properties including beam energy and phase while actuators control the incident wave of the klystron and individual cavity resonance frequencies. The operability of a large llrf system requires a high degree of automation while the high availability requires robust algorithms, redundancy, and extremel...

  7. Further linear algebra

    CERN Document Server

    Blyth, T S

    2002-01-01

    Most of the introductory courses on linear algebra develop the basic theory of finite­ dimensional vector spaces, and in so doing relate the notion of a linear mapping to that of a matrix. Generally speaking, such courses culminate in the diagonalisation of certain matrices and the application of this process to various situations. Such is the case, for example, in our previous SUMS volume Basic Linear Algebra. The present text is a continuation of that volume, and has the objective of introducing the reader to more advanced properties of vector spaces and linear mappings, and consequently of matrices. For readers who are not familiar with the contents of Basic Linear Algebra we provide an introductory chapter that consists of a compact summary of the prerequisites for the present volume. In order to consolidate the student's understanding we have included a large num­ ber of illustrative and worked examples, as well as many exercises that are strategi­ cally placed throughout the text. Solutions to the ex...

  8. 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

  9. Linear mass reflectron

    International Nuclear Information System (INIS)

    Mamyrin, B.A.; Shmikk, D.V.

    1979-01-01

    A description and operating principle of a linear mass reflectron with V-form trajectory of ion motion -a new non-magnetic time-of-flight mass spectrometer with high resolution are presented. The ion-optical system of the device consists of an ion source with ionization by electron shock, of accelerating gaps, reflector gaps, a drift space and ion detector. Ions move in the linear mass refraction along the trajectories parallel to the axis of the analyzer chamber. The results of investigations into the experimental device are given. With an ion drift length of 0.6 m the device resolution is 1200 with respect to the peak width at half-height. Small-sized mass spectrometric transducers with high resolution and sensitivity may be designed on the base of the linear mass reflectron principle

  10. Applied linear algebra

    CERN Document Server

    Olver, Peter J

    2018-01-01

    This textbook develops the essential tools of linear algebra, with the goal of imparting technique alongside contextual understanding. Applications go hand-in-hand with theory, each reinforcing and explaining the other. This approach encourages students to develop not only the technical proficiency needed to go on to further study, but an appreciation for when, why, and how the tools of linear algebra can be used across modern applied mathematics. Providing an extensive treatment of essential topics such as Gaussian elimination, inner products and norms, and eigenvalues and singular values, this text can be used for an in-depth first course, or an application-driven second course in linear algebra. In this second edition, applications have been updated and expanded to include numerical methods, dynamical systems, data analysis, and signal processing, while the pedagogical flow of the core material has been improved. Throughout, the text emphasizes the conceptual connections between each application and the un...

  11. Theory of linear operations

    CERN Document Server

    Banach, S

    1987-01-01

    This classic work by the late Stefan Banach has been translated into English so as to reach a yet wider audience. It contains the basics of the algebra of operators, concentrating on the study of linear operators, which corresponds to that of the linear forms a1x1 + a2x2 + ... + anxn of algebra.The book gathers results concerning linear operators defined in general spaces of a certain kind, principally in Banach spaces, examples of which are: the space of continuous functions, that of the pth-power-summable functions, Hilbert space, etc. The general theorems are interpreted in various mathematical areas, such as group theory, differential equations, integral equations, equations with infinitely many unknowns, functions of a real variable, summation methods and orthogonal series.A new fifty-page section (``Some Aspects of the Present Theory of Banach Spaces'''') complements this important monograph.

  12. Dimension of linear models

    DEFF Research Database (Denmark)

    Høskuldsson, Agnar

    1996-01-01

    Determination of the proper dimension of a given linear model is one of the most important tasks in the applied modeling work. We consider here eight criteria that can be used to determine the dimension of the model, or equivalently, the number of components to use in the model. Four of these cri......Determination of the proper dimension of a given linear model is one of the most important tasks in the applied modeling work. We consider here eight criteria that can be used to determine the dimension of the model, or equivalently, the number of components to use in the model. Four...... the basic problems in determining the dimension of linear models. Then each of the eight measures are treated. The results are illustrated by examples....

  13. Linear programming using Matlab

    CERN Document Server

    Ploskas, Nikolaos

    2017-01-01

    This book offers a theoretical and computational presentation of a variety of linear programming algorithms and methods with an emphasis on the revised simplex method and its components. A theoretical background and mathematical formulation is included for each algorithm as well as comprehensive numerical examples and corresponding MATLAB® code. The MATLAB® implementations presented in this book  are sophisticated and allow users to find solutions to large-scale benchmark linear programs. Each algorithm is followed by a computational study on benchmark problems that analyze the computational behavior of the presented algorithms. As a solid companion to existing algorithmic-specific literature, this book will be useful to researchers, scientists, mathematical programmers, and students with a basic knowledge of linear algebra and calculus.  The clear presentation enables the reader to understand and utilize all components of simplex-type methods, such as presolve techniques, scaling techniques, pivoting ru...

  14. Linear Colliders TESLA

    International Nuclear Information System (INIS)

    Anon.

    1994-01-01

    The aim of the TESLA (TeV Superconducting Linear Accelerator) collaboration (at present 19 institutions from seven countries) is to establish the technology for a high energy electron-positron linear collider using superconducting radiofrequency cavities to accelerate its beams. Another basic goal is to demonstrate that such a collider can meet its performance goals in a cost effective manner. For this the TESLA collaboration is preparing a 500 MeV superconducting linear test accelerator at the DESY Laboratory in Hamburg. This TTF (TESLA Test Facility) consists of four cryomodules, each approximately 12 m long and containing eight 9-cell solid niobium cavities operating at a frequency of 1.3 GHz

  15. Linearly Adjustable International Portfolios

    Science.gov (United States)

    Fonseca, R. J.; Kuhn, D.; Rustem, B.

    2010-09-01

    We present an approach to multi-stage international portfolio optimization based on the imposition of a linear structure on the recourse decisions. Multiperiod decision problems are traditionally formulated as stochastic programs. Scenario tree based solutions however can become intractable as the number of stages increases. By restricting the space of decision policies to linear rules, we obtain a conservative tractable approximation to the original problem. Local asset prices and foreign exchange rates are modelled separately, which allows for a direct measure of their impact on the final portfolio value.

  16. Linearly Adjustable International Portfolios

    International Nuclear Information System (INIS)

    Fonseca, R. J.; Kuhn, D.; Rustem, B.

    2010-01-01

    We present an approach to multi-stage international portfolio optimization based on the imposition of a linear structure on the recourse decisions. Multiperiod decision problems are traditionally formulated as stochastic programs. Scenario tree based solutions however can become intractable as the number of stages increases. By restricting the space of decision policies to linear rules, we obtain a conservative tractable approximation to the original problem. Local asset prices and foreign exchange rates are modelled separately, which allows for a direct measure of their impact on the final portfolio value.

  17. Linear induction motor

    International Nuclear Information System (INIS)

    Barkman, W.E.; Adams, W.Q.; Berrier, B.R.

    1978-01-01

    A linear induction motor has been operated on a test bed with a feedback pulse resolution of 5 nm (0.2 μin). Slewing tests with this slide drive have shown positioning errors less than or equal to 33 nm (1.3 μin) at feedrates between 0 and 25.4 mm/min (0-1 ipm). A 0.86-m (34-in)-stroke linear motor is being investigated, using the SPACO machine as a test bed. Initial results were encouraging, and work is continuing to optimize the servosystem compensation

  18. Handbook of linear algebra

    CERN Document Server

    Hogben, Leslie

    2013-01-01

    With a substantial amount of new material, the Handbook of Linear Algebra, Second Edition provides comprehensive coverage of linear algebra concepts, applications, and computational software packages in an easy-to-use format. It guides you from the very elementary aspects of the subject to the frontiers of current research. Along with revisions and updates throughout, the second edition of this bestseller includes 20 new chapters.New to the Second EditionSeparate chapters on Schur complements, additional types of canonical forms, tensors, matrix polynomials, matrix equations, special types of

  19. Linear Algebra Thoroughly Explained

    CERN Document Server

    Vujičić, Milan

    2008-01-01

    Linear Algebra Thoroughly Explained provides a comprehensive introduction to the subject suitable for adoption as a self-contained text for courses at undergraduate and postgraduate level. The clear and comprehensive presentation of the basic theory is illustrated throughout with an abundance of worked examples. The book is written for teachers and students of linear algebra at all levels and across mathematics and the applied sciences, particularly physics and engineering. It will also be an invaluable addition to research libraries as a comprehensive resource book for the subject.

  20. America, Linearly Cyclical

    Science.gov (United States)

    2013-05-10

    AND VICTIM- ~ vAP BLAMING 4. AMERICA, LINEARLY CYCUCAL AF IMT 1768, 19840901, V5 PREVIOUS EDITION WILL BE USED. C2C Jessica Adams Dr. Brissett...his desires, his failings, and his aspirations follow the same general trend throughout history and throughout cultures. The founding fathers sought

  1. Stanford's linear collider

    International Nuclear Information System (INIS)

    Southworth, B.

    1985-01-01

    The peak of the construction phase of the Stanford Linear Collider, SLC, to achieve 50 GeV electron-positron collisions has now been passed. The work remains on schedule to attempt colliding beams, initially at comparatively low luminosity, early in 1987. (orig./HSI).

  2. Dosimetry of linear sources

    International Nuclear Information System (INIS)

    Mafra Neto, F.

    1992-01-01

    The dose of gamma radiation from a linear source of cesium 137 is obtained, presenting two difficulties: oblique filtration of radiation when cross the platinum wall, in different directions, and dose connection due to the scattering by the material mean of propagation. (C.G.C.)

  3. Resistors Improve Ramp Linearity

    Science.gov (United States)

    Kleinberg, L. L.

    1982-01-01

    Simple modification to bootstrap ramp generator gives more linear output over longer sweep times. New circuit adds just two resistors, one of which is adjustable. Modification cancels nonlinearities due to variations in load on charging capacitor and due to changes in charging current as the voltage across capacitor increases.

  4. LINEAR COLLIDERS: 1992 workshop

    International Nuclear Information System (INIS)

    Settles, Ron; Coignet, Guy

    1992-01-01

    As work on designs for future electron-positron linear colliders pushes ahead at major Laboratories throughout the world in a major international collaboration framework, the LC92 workshop held in Garmisch Partenkirchen this summer, attended by 200 machine and particle physicists, provided a timely focus

  5. Linear genetic programming

    CERN Document Server

    Brameier, Markus

    2007-01-01

    Presents a variant of Genetic Programming that evolves imperative computer programs as linear sequences of instructions, in contrast to the more traditional functional expressions or syntax trees. This book serves as a reference for researchers, but also contains sufficient introduction for students and those who are new to the field

  6. On Solving Linear Recurrences

    Science.gov (United States)

    Dobbs, David E.

    2013-01-01

    A direct method is given for solving first-order linear recurrences with constant coefficients. The limiting value of that solution is studied as "n to infinity." This classroom note could serve as enrichment material for the typical introductory course on discrete mathematics that follows a calculus course.

  7. Review of linear colliders

    International Nuclear Information System (INIS)

    Takeda, Seishi

    1992-01-01

    The status of R and D of future e + e - linear colliders proposed by the institutions throughout the world is described including the JLC, NLC, VLEPP, CLIC, DESY/THD and TESLA projects. The parameters and RF sources are discussed. (G.P.) 36 refs.; 1 tab

  8. Finite-dimensional linear algebra

    CERN Document Server

    Gockenbach, Mark S

    2010-01-01

    Some Problems Posed on Vector SpacesLinear equationsBest approximationDiagonalizationSummaryFields and Vector SpacesFields Vector spaces Subspaces Linear combinations and spanning sets Linear independence Basis and dimension Properties of bases Polynomial interpolation and the Lagrange basis Continuous piecewise polynomial functionsLinear OperatorsLinear operatorsMore properties of linear operatorsIsomorphic vector spaces Linear operator equations Existence and uniqueness of solutions The fundamental theorem; inverse operatorsGaussian elimination Newton's method Linear ordinary differential eq

  9. Linearity and Non-linearity of Photorefractive effect in Materials ...

    African Journals Online (AJOL)

    In this paper we have studied the Linearity and Non-linearity of Photorefractive effect in materials using the band transport model. For low light beam intensities the change in the refractive index is proportional to the electric field for linear optics while for non- linear optics the change in refractive index is directly proportional ...

  10. Linearly Refined Session Types

    Directory of Open Access Journals (Sweden)

    Pedro Baltazar

    2012-11-01

    Full Text Available Session types capture precise protocol structure in concurrent programming, but do not specify properties of the exchanged values beyond their basic type. Refinement types are a form of dependent types that can address this limitation, combining types with logical formulae that may refer to program values and can constrain types using arbitrary predicates. We present a pi calculus with assume and assert operations, typed using a session discipline that incorporates refinement formulae written in a fragment of Multiplicative Linear Logic. Our original combination of session and refinement types, together with the well established benefits of linearity, allows very fine-grained specifications of communication protocols in which refinement formulae are treated as logical resources rather than persistent truths.

  11. Linear Water Waves

    Science.gov (United States)

    Kuznetsov, N.; Maz'ya, V.; Vainberg, B.

    2002-08-01

    This book gives a self-contained and up-to-date account of mathematical results in the linear theory of water waves. The study of waves has many applications, including the prediction of behavior of floating bodies (ships, submarines, tension-leg platforms etc.), the calculation of wave-making resistance in naval architecture, and the description of wave patterns over bottom topography in geophysical hydrodynamics. The first section deals with time-harmonic waves. Three linear boundary value problems serve as the approximate mathematical models for these types of water waves. The next section uses a plethora of mathematical techniques in the investigation of these three problems. The techniques used in the book include integral equations based on Green's functions, various inequalities between the kinetic and potential energy and integral identities which are indispensable for proving the uniqueness theorems. The so-called inverse procedure is applied to constructing examples of non-uniqueness, usually referred to as 'trapped nodes.'

  12. The International Linear Collider

    Directory of Open Access Journals (Sweden)

    List Benno

    2014-04-01

    Full Text Available The International Linear Collider (ILC is a proposed e+e− linear collider with a centre-of-mass energy of 200–500 GeV, based on superconducting RF cavities. The ILC would be an ideal machine for precision studies of a light Higgs boson and the top quark, and would have a discovery potential for new particles that is complementary to that of LHC. The clean experimental conditions would allow the operation of detectors with extremely good performance; two such detectors, ILD and SiD, are currently being designed. Both make use of novel concepts for tracking and calorimetry. The Japanese High Energy Physics community has recently recommended to build the ILC in Japan.

  13. The International Linear Collider

    Science.gov (United States)

    List, Benno

    2014-04-01

    The International Linear Collider (ILC) is a proposed e+e- linear collider with a centre-of-mass energy of 200-500 GeV, based on superconducting RF cavities. The ILC would be an ideal machine for precision studies of a light Higgs boson and the top quark, and would have a discovery potential for new particles that is complementary to that of LHC. The clean experimental conditions would allow the operation of detectors with extremely good performance; two such detectors, ILD and SiD, are currently being designed. Both make use of novel concepts for tracking and calorimetry. The Japanese High Energy Physics community has recently recommended to build the ILC in Japan.

  14. Dimension of linear models

    DEFF Research Database (Denmark)

    Høskuldsson, Agnar

    1996-01-01

    Determination of the proper dimension of a given linear model is one of the most important tasks in the applied modeling work. We consider here eight criteria that can be used to determine the dimension of the model, or equivalently, the number of components to use in the model. Four...... the basic problems in determining the dimension of linear models. Then each of the eight measures are treated. The results are illustrated by examples....... of these criteria are widely used ones, while the remaining four are ones derived from the H-principle of mathematical modeling. Many examples from practice show that the criteria derived from the H-principle function better than the known and popular criteria for the number of components. We shall briefly review...

  15. Reciprocating linear motor

    Science.gov (United States)

    Goldowsky, Michael P. (Inventor)

    1987-01-01

    A reciprocating linear motor is formed with a pair of ring-shaped permanent magnets having opposite radial polarizations, held axially apart by a nonmagnetic yoke, which serves as an axially displaceable armature assembly. A pair of annularly wound coils having axial lengths which differ from the axial lengths of the permanent magnets are serially coupled together in mutual opposition and positioned with an outer cylindrical core in axial symmetry about the armature assembly. One embodiment includes a second pair of annularly wound coils serially coupled together in mutual opposition and an inner cylindrical core positioned in axial symmetry inside the armature radially opposite to the first pair of coils. Application of a potential difference across a serial connection of the two pairs of coils creates a current flow perpendicular to the magnetic field created by the armature magnets, thereby causing limited linear displacement of the magnets relative to the coils.

  16. Duality in linearized gravity

    International Nuclear Information System (INIS)

    Henneaux, Marc; Teitelboim, Claudio

    2005-01-01

    We show that duality transformations of linearized gravity in four dimensions, i.e., rotations of the linearized Riemann tensor and its dual into each other, can be extended to the dynamical fields of the theory so as to be symmetries of the action and not just symmetries of the equations of motion. Our approach relies on the introduction of two superpotentials, one for the spatial components of the spin-2 field and the other for their canonically conjugate momenta. These superpotentials are two-index, symmetric tensors. They can be taken to be the basic dynamical fields and appear locally in the action. They are simply rotated into each other under duality. In terms of the superpotentials, the canonical generator of duality rotations is found to have a Chern-Simons-like structure, as in the Maxwell case

  17. The SLAC linear collider

    International Nuclear Information System (INIS)

    Phinney, N.

    1992-01-01

    The SLAC Linear Collider has begun a new era of operation with the SLD detector. During 1991 there was a first engineering run for the SLD in parallel with machine improvements to increase luminosity and reliability. For the 1992 run, a polarized electron source was added and more than 10,000 Zs with an average of 23% polarization have been logged by the SLD. This paper discusses the performance of the SLC in 1991 and 1992 and the technical advances that have produced higher luminosity. Emphasis will be placed on issues relevant to future linear colliders such as producing and maintaining high current, low emittance beams and focusing the beams to the micron scale for collisions. (Author) tab., 2 figs., 18 refs

  18. Linear waves and instabilities

    International Nuclear Information System (INIS)

    Bers, A.

    1975-01-01

    The electrodynamic equations for small-amplitude waves and their dispersion relation in a homogeneous plasma are outlined. For such waves, energy and momentum, and their flow and transformation, are described. Perturbation theory of waves is treated and applied to linear coupling of waves, and the resulting instabilities from such interactions between active and passive waves. Linear stability analysis in time and space is described where the time-asymptotic, time-space Green's function for an arbitrary dispersion relation is developed. The perturbation theory of waves is applied to nonlinear coupling, with particular emphasis on pump-driven interactions of waves. Details of the time--space evolution of instabilities due to coupling are given. (U.S.)

  19. Extended linear chain compounds

    CERN Document Server

    Linear chain substances span a large cross section of contemporary chemistry ranging from covalent polymers, to organic charge transfer com­ plexes to nonstoichiometric transition metal coordination complexes. Their commonality, which coalesced intense interest in the theoretical and exper­ imental solid state physics/chemistry communities, was based on the obser­ vation that these inorganic and organic polymeric substrates exhibit striking metal-like electrical and optical properties. Exploitation and extension of these systems has led to the systematic study of both the chemistry and physics of highly and poorly conducting linear chain substances. To gain a salient understanding of these complex materials rich in anomalous aniso­ tropic electrical, optical, magnetic, and mechanical properties, the conver­ gence of diverse skills and talents was required. The constructive blending of traditionally segregated disciplines such as synthetic and physical organic, inorganic, and polymer chemistry, crystallog...

  20. Non-linear osmosis

    Science.gov (United States)

    Diamond, Jared M.

    1966-01-01

    1. The relation between osmotic gradient and rate of osmotic water flow has been measured in rabbit gall-bladder by a gravimetric procedure and by a rapid method based on streaming potentials. Streaming potentials were directly proportional to gravimetrically measured water fluxes. 2. As in many other tissues, water flow was found to vary with gradient in a markedly non-linear fashion. There was no consistent relation between the water permeability and either the direction or the rate of water flow. 3. Water flow in response to a given gradient decreased at higher osmolarities. The resistance to water flow increased linearly with osmolarity over the range 186-825 m-osM. 4. The resistance to water flow was the same when the gall-bladder separated any two bathing solutions with the same average osmolarity, regardless of the magnitude of the gradient. In other words, the rate of water flow is given by the expression (Om — Os)/[Ro′ + ½k′ (Om + Os)], where Ro′ and k′ are constants and Om and Os are the bathing solution osmolarities. 5. Of the theories advanced to explain non-linear osmosis in other tissues, flow-induced membrane deformations, unstirred layers, asymmetrical series-membrane effects, and non-osmotic effects of solutes could not explain the results. However, experimental measurements of water permeability as a function of osmolarity permitted quantitative reconstruction of the observed water flow—osmotic gradient curves. Hence non-linear osmosis in rabbit gall-bladder is due to a decrease in water permeability with increasing osmolarity. 6. The results suggest that aqueous channels in the cell membrane behave as osmometers, shrinking in concentrated solutions of impermeant molecules and thereby increasing membrane resistance to water flow. A mathematical formulation of such a membrane structure is offered. PMID:5945254

  1. Fundamentals of linear algebra

    CERN Document Server

    Dash, Rajani Ballav

    2008-01-01

    FUNDAMENTALS OF LINEAR ALGEBRA is a comprehensive Text Book, which can be used by students and teachers of All Indian Universities. The Text has easy, understandable form and covers all topics of UGC Curriculum. There are lots of worked out examples which helps the students in solving the problems without anybody's help. The Problem sets have been designed keeping in view of the questions asked in different examinations.

  2. Linear network theory

    CERN Document Server

    Sander, K F

    1964-01-01

    Linear Network Theory covers the significant algebraic aspect of network theory, with minimal reference to practical circuits. The book begins the presentation of network analysis with the exposition of networks containing resistances only, and follows it up with a discussion of networks involving inductance and capacity by way of the differential equations. Classification and description of certain networks, equivalent networks, filter circuits, and network functions are also covered. Electrical engineers, technicians, electronics engineers, electricians, and students learning the intricacies

  3. Non linear viscoelastic models

    DEFF Research Database (Denmark)

    Agerkvist, Finn T.

    2011-01-01

    Viscoelastic eects are often present in loudspeaker suspensions, this can be seen in the displacement transfer function which often shows a frequency dependent value below the resonance frequency. In this paper nonlinear versions of the standard linear solid model (SLS) are investigated....... The simulations show that the nonlinear version of the Maxwell SLS model can result in a time dependent small signal stiness while the Kelvin Voight version does not....

  4. Relativistic Linear Restoring Force

    Science.gov (United States)

    Clark, D.; Franklin, J.; Mann, N.

    2012-01-01

    We consider two different forms for a relativistic version of a linear restoring force. The pair comes from taking Hooke's law to be the force appearing on the right-hand side of the relativistic expressions: d"p"/d"t" or d"p"/d["tau"]. Either formulation recovers Hooke's law in the non-relativistic limit. In addition to these two forces, we…

  5. Superconducting linear colliders

    International Nuclear Information System (INIS)

    Anon.

    1990-01-01

    The advantages of superconducting radiofrequency (SRF) for particle accelerators have been demonstrated by successful operation of systems in the TRISTAN and LEP electron-positron collider rings respectively at the Japanese KEK Laboratory and at CERN. If performance continues to improve and costs can be lowered, this would open an attractive option for a high luminosity TeV (1000 GeV) linear collider

  6. Perturbed asymptotically linear problems

    OpenAIRE

    Bartolo, R.; Candela, A. M.; Salvatore, A.

    2012-01-01

    The aim of this paper is investigating the existence of solutions of some semilinear elliptic problems on open bounded domains when the nonlinearity is subcritical and asymptotically linear at infinity and there is a perturbation term which is just continuous. Also in the case when the problem has not a variational structure, suitable procedures and estimates allow us to prove that the number of distinct crtitical levels of the functional associated to the unperturbed problem is "stable" unde...

  7. Miniature linear cooler development

    International Nuclear Information System (INIS)

    Pruitt, G.R.

    1993-01-01

    An overview is presented of the status of a family of miniature linear coolers currently under development by Hughes Aircraft Co. for use in hand held, volume limited or power limited infrared applications. These coolers, representing the latest additions to the Hughes family of TOP trademark [twin-opposed piston] linear coolers, have been fabricated and tested in three different configurations. Each configuration is designed to utilize a common compressor assembly resulting in reduced manufacturing costs. The baseline compressor has been integrated with two different expander configurations and has been operated with two different levels of input power. These various configuration combinations offer a wide range of performance and interface characteristics which may be tailored to applications requiring limited power and size without significantly compromising cooler capacity or cooldown characteristics. Key cooler characteristics and test data are summarized for three combinations of cooler configurations which are representative of the versatility of this linear cooler design. Configurations reviewed include the shortened coldfinger [1.50 to 1.75 inches long], limited input power [less than 17 Watts] for low power availability applications; the shortened coldfinger with higher input power for lightweight, higher performance applications; and coldfingers compatible with DoD 0.4 Watt Common Module coolers for wider range retrofit capability. Typical weight of these miniature linear coolers is less than 500 grams for the compressor, expander and interconnecting transfer line. Cooling capacity at 80K at room ambient conditions ranges from 400 mW to greater than 550 mW. Steady state power requirements for maintaining a heat load of 150 mW at 80K has been shown to be less than 8 Watts. Ongoing reliability growth testing is summarized including a review of the latest test article results

  8. Linear pneumatic actuator

    Directory of Open Access Journals (Sweden)

    Avram Mihai

    2017-01-01

    Full Text Available The paper presents a linear pneumatic actuator with short working stroke. It consists of a pneumatic motor (a simple stroke cylinder or a membrane chamber, two 2/2 pneumatic distributors “all or nothing” electrically commanded for controlling the intake/outtake flow to/from the active chamber of the motor, a position transducer and a microcontroller. There is also presented the theoretical analysis (mathematical modelling and numerical simulation accomplished.

  9. Linear pneumatic actuator

    OpenAIRE

    Avram Mihai; Niţu Constantin; Bucşan Constantin; Grămescu Bogdan

    2017-01-01

    The paper presents a linear pneumatic actuator with short working stroke. It consists of a pneumatic motor (a simple stroke cylinder or a membrane chamber), two 2/2 pneumatic distributors “all or nothing” electrically commanded for controlling the intake/outtake flow to/from the active chamber of the motor, a position transducer and a microcontroller. There is also presented the theoretical analysis (mathematical modelling and numerical simulation) accomplished.

  10. Linear MHD equilibria

    International Nuclear Information System (INIS)

    Scheffel, J.

    1984-03-01

    The linear Grad-Shafranov equation for a toroidal, axisymmetric plasma is solved analytically. Exact solutions are given in terms of confluent hyper-geometric functions. As an alternative, simple and accurate WKBJ solutions are presented. With parabolic pressure profiles, both hollow and peaked toroidal current density profiles are obtained. As an example the equilibrium of a z-pinch with a square-shaped cross section is derived.(author)

  11. Linear induction accelerator

    Science.gov (United States)

    Buttram, M.T.; Ginn, J.W.

    1988-06-21

    A linear induction accelerator includes a plurality of adder cavities arranged in a series and provided in a structure which is evacuated so that a vacuum inductance is provided between each adder cavity and the structure. An energy storage system for the adder cavities includes a pulsed current source and a respective plurality of bipolar converting networks connected thereto. The bipolar high-voltage, high-repetition-rate square pulse train sets and resets the cavities. 4 figs.

  12. Linear algebraic groups

    CERN Document Server

    Springer, T A

    1998-01-01

    "[The first] ten chapters...are an efficient, accessible, and self-contained introduction to affine algebraic groups over an algebraically closed field. The author includes exercises and the book is certainly usable by graduate students as a text or for self-study...the author [has a] student-friendly style… [The following] seven chapters... would also be a good introduction to rationality issues for algebraic groups. A number of results from the literature…appear for the first time in a text." –Mathematical Reviews (Review of the Second Edition) "This book is a completely new version of the first edition. The aim of the old book was to present the theory of linear algebraic groups over an algebraically closed field. Reading that book, many people entered the research field of linear algebraic groups. The present book has a wider scope. Its aim is to treat the theory of linear algebraic groups over arbitrary fields. Again, the author keeps the treatment of prerequisites self-contained. The material of t...

  13. Parametric Linear Dynamic Logic

    Directory of Open Access Journals (Sweden)

    Peter Faymonville

    2014-08-01

    Full Text Available We introduce Parametric Linear Dynamic Logic (PLDL, which extends Linear Dynamic Logic (LDL by temporal operators equipped with parameters that bound their scope. LDL was proposed as an extension of Linear Temporal Logic (LTL that is able to express all ω-regular specifications while still maintaining many of LTL's desirable properties like an intuitive syntax and a translation into non-deterministic Büchi automata of exponential size. But LDL lacks capabilities to express timing constraints. By adding parameterized operators to LDL, we obtain a logic that is able to express all ω-regular properties and that subsumes parameterized extensions of LTL like Parametric LTL and PROMPT-LTL. Our main technical contribution is a translation of PLDL formulas into non-deterministic Büchi word automata of exponential size via alternating automata. This yields a PSPACE model checking algorithm and a realizability algorithm with doubly-exponential running time. Furthermore, we give tight upper and lower bounds on optimal parameter values for both problems. These results show that PLDL model checking and realizability are not harder than LTL model checking and realizability.

  14. Quantum linear Boltzmann equation

    International Nuclear Information System (INIS)

    Vacchini, Bassano; Hornberger, Klaus

    2009-01-01

    We review the quantum version of the linear Boltzmann equation, which describes in a non-perturbative fashion, by means of scattering theory, how the quantum motion of a single test particle is affected by collisions with an ideal background gas. A heuristic derivation of this Lindblad master equation is presented, based on the requirement of translation-covariance and on the relation to the classical linear Boltzmann equation. After analyzing its general symmetry properties and the associated relaxation dynamics, we discuss a quantum Monte Carlo method for its numerical solution. We then review important limiting forms of the quantum linear Boltzmann equation, such as the case of quantum Brownian motion and pure collisional decoherence, as well as the application to matter wave optics. Finally, we point to the incorporation of quantum degeneracies and self-interactions in the gas by relating the equation to the dynamic structure factor of the ambient medium, and we provide an extension of the equation to include internal degrees of freedom.

  15. The Stanford Linear Collider

    International Nuclear Information System (INIS)

    Emma, P.

    1995-01-01

    The Stanford Linear Collider (SLC) is the first and only high-energy e + e - linear collider in the world. Its most remarkable features are high intensity, submicron sized, polarized (e - ) beams at a single interaction point. The main challenges posed by these unique characteristics include machine-wide emittance preservation, consistent high intensity operation, polarized electron production and transport, and the achievement of a high degree of beam stability on all time scales. In addition to serving as an important machine for the study of Z 0 boson production and decay using polarized beams, the SLC is also an indispensable source of hands-on experience for future linear colliders. Each new year of operation has been highlighted with a marked improvement in performance. The most significant improvements for the 1994-95 run include new low impedance vacuum chambers for the damping rings, an upgrade to the optics and diagnostics of the final focus systems, and a higher degree of polarization from the electron source. As a result, the average luminosity has nearly doubled over the previous year with peaks approaching 10 30 cm -2 s -1 and an 80% electron polarization at the interaction point. These developments as well as the remaining identifiable performance limitations will be discussed

  16. Non linear microtearing modes

    International Nuclear Information System (INIS)

    Garbet, X.; Mourgues, F.; Samain, A.

    1987-01-01

    Among the various instabilities which could explain the anomalous electron heat transport observed in tokamaks during additional heating, a microtearing turbulence is a reasonable candidate since it affects directly the magnetic topology. This turbulence may be described in a proper frame rotating around the majors axis by a static potential vector. In strong non linear regimes, the flow of electrons along the stochastic field lines induces a current. The point is to know whether this current can sustain the turbulence. The mechanisms of this self-consistency, involving the combined effects of the thermal diamagnetism and of the electric drift are presented here

  17. RF linear accelerators

    CERN Document Server

    Wangler, Thomas P

    2008-01-01

    Thomas P. Wangler received his B.S. degree in physics from Michigan State University, and his Ph.D. degree in physics and astronomy from the University of Wisconsin. After postdoctoral appointments at the University of Wisconsin and Brookhaven National Laboratory, he joined the staff of Argonne National Laboratory in 1966, working in the fields of experimental high-energy physics and accelerator physics. He joined the Accelerator Technology Division at Los Alamos National Laboratory in 1979, where he specialized in high-current beam physics and linear accelerator design and technology. In 2007

  18. SLAC linear collider

    International Nuclear Information System (INIS)

    Richter, B.; Bell, R.A.; Brown, K.L.

    1980-06-01

    The SLAC LINEAR COLLIDER is designed to achieve an energy of 100 GeV in the electron-positron center-of-mass system by accelerating intense bunches of particles in the SLAC linac and transporting the electron and positron bunches in a special magnet system to a point where they are focused to a radius of about 2 microns and made to collide head on. The rationale for this new type of colliding beam system is discussed, the project is described, some of the novel accelerator physics issues involved are discussed, and some of the critical technical components are described

  19. Matlab linear algebra

    CERN Document Server

    Lopez, Cesar

    2014-01-01

    MATLAB is a high-level language and environment for numerical computation, visualization, and programming. Using MATLAB, you can analyze data, develop algorithms, and create models and applications. The language, tools, and built-in math functions enable you to explore multiple approaches and reach a solution faster than with spreadsheets or traditional programming languages, such as C/C++ or Java. MATLAB Linear Algebra introduces you to the MATLAB language with practical hands-on instructions and results, allowing you to quickly achieve your goals. In addition to giving an introduction to

  20. Special set linear algebra and special set fuzzy linear algebra

    OpenAIRE

    Kandasamy, W. B. Vasantha; Smarandache, Florentin; Ilanthenral, K.

    2009-01-01

    The authors in this book introduce the notion of special set linear algebra and special set fuzzy Linear algebra, which is an extension of the notion set linear algebra and set fuzzy linear algebra. These concepts are best suited in the application of multi expert models and cryptology. This book has five chapters. In chapter one the basic concepts about set linear algebra is given in order to make this book a self contained one. The notion of special set linear algebra and their fuzzy analog...

  1. Electrodynamic linear motor

    Energy Technology Data Exchange (ETDEWEB)

    Munehiro, H

    1980-05-29

    When driving the carriage of a printer through a rotating motor, there are problems regarding the limited accuracy of the carriage position due to rotation or contraction and ageing of the cable. In order to solve the problem, a direct drive system was proposed, in which the printer carriage is driven by a linear motor. If one wants to keep the motor circuit of such a motor compact, then the magnetic flux density in the air gap must be reduced or the motor travel must be reduced. It is the purpose of this invention to create an electrodynamic linear motor, which on the one hand is compact and light and on the other hand has a relatively high constant force over a large travel. The invention is characterised by the fact that magnetic fields of alternating polarity are generated at equal intervals in the magnetic field, and that the coil arrangement has 2 adjacent coils, whose size corresponds to half the length of each magnetic pole. A logic circuit is provided to select one of the two coils and to determine the direction of the current depending on the signals of a magnetic field sensor on the coil arrangement.

  2. Linear wind generator

    International Nuclear Information System (INIS)

    Kozarov, A.; Petrov, O.; Antonov, J.; Sotirova, S.; Petrova, B.

    2006-01-01

    The purpose of the linear wind-power generator described in this article is to decrease the following disadvantages of the common wind-powered turbine: 1) large bending and twisting moments to the blades and the shaft, especially when strong winds and turbulence exist; 2) significant values of the natural oscillation period of the construction result in the possibility of occurrence of destroying resonance oscillations; 3) high velocity of the peripheral parts of the rotor creating a danger for birds; 4) difficulties, connected with the installation and the operation on the mountain ridges and passages where the wind energy potential is the largest. The working surfaces of the generator in questions driven by the wind are not connected with a joint shaft but each moves along a railway track with few oscillations. So the sizes of each component are small and their number can be rather large. The mechanical trajectory is not a circle but a closed outline in a vertical plain, which consists of two rectilinear sectors, one above the other, connected in their ends by semi-circumferences. The mechanical energy of each component turns into electrical on the principle of the linear electrical generator. A regulation is provided when the direction of the wind is perpendicular to the route. A possibility of effectiveness is shown through aiming of additional quantities of air to the movable components by static barriers

  3. Linearization of the Lorenz system

    International Nuclear Information System (INIS)

    Li, Chunbiao; Sprott, Julien Clinton; Thio, Wesley

    2015-01-01

    A partial and complete piecewise linearized version of the Lorenz system is proposed. The linearized versions have an independent total amplitude control parameter. Additional further linearization leads naturally to a piecewise linear version of the diffusionless Lorenz system. A chaotic circuit with a single amplitude controller is then implemented using a new switch element, producing a chaotic oscillation that agrees with the numerical calculation for the piecewise linear diffusionless Lorenz system. - Highlights: • A partial and complete piecewise linearized version of the Lorenz system are addressed. • The linearized versions have an independent total amplitude control parameter. • A piecewise linear version of the diffusionless Lorenz system is derived by further linearization. • A corresponding chaotic circuit without any multiplier is implemented for the chaotic oscillation

  4. Topics in computational linear optimization

    DEFF Research Database (Denmark)

    Hultberg, Tim Helge

    2000-01-01

    Linear optimization has been an active area of research ever since the pioneering work of G. Dantzig more than 50 years ago. This research has produced a long sequence of practical as well as theoretical improvements of the solution techniques avilable for solving linear optimization problems...... of high quality solvers and the use of algebraic modelling systems to handle the communication between the modeller and the solver. This dissertation features four topics in computational linear optimization: A) automatic reformulation of mixed 0/1 linear programs, B) direct solution of sparse unsymmetric...... systems of linear equations, C) reduction of linear programs and D) integration of algebraic modelling of linear optimization problems in C++. Each of these topics is treated in a separate paper included in this dissertation. The efficiency of solving mixed 0-1 linear programs by linear programming based...

  5. Linearization of the Lorenz system

    Energy Technology Data Exchange (ETDEWEB)

    Li, Chunbiao, E-mail: goontry@126.com [School of Electronic & Information Engineering, Nanjing University of Information Science & Technology, Nanjing 210044 (China); Engineering Technology Research and Development Center of Jiangsu Circulation Modernization Sensor Network, Jiangsu Institute of Commerce, Nanjing 211168 (China); Sprott, Julien Clinton [Department of Physics, University of Wisconsin–Madison, Madison, WI 53706 (United States); Thio, Wesley [Department of Electrical and Computer Engineering, The Ohio State University, Columbus, OH 43210 (United States)

    2015-05-08

    A partial and complete piecewise linearized version of the Lorenz system is proposed. The linearized versions have an independent total amplitude control parameter. Additional further linearization leads naturally to a piecewise linear version of the diffusionless Lorenz system. A chaotic circuit with a single amplitude controller is then implemented using a new switch element, producing a chaotic oscillation that agrees with the numerical calculation for the piecewise linear diffusionless Lorenz system. - Highlights: • A partial and complete piecewise linearized version of the Lorenz system are addressed. • The linearized versions have an independent total amplitude control parameter. • A piecewise linear version of the diffusionless Lorenz system is derived by further linearization. • A corresponding chaotic circuit without any multiplier is implemented for the chaotic oscillation.

  6. On the linear programming bound for linear Lee codes.

    Science.gov (United States)

    Astola, Helena; Tabus, Ioan

    2016-01-01

    Based on an invariance-type property of the Lee-compositions of a linear Lee code, additional equality constraints can be introduced to the linear programming problem of linear Lee codes. In this paper, we formulate this property in terms of an action of the multiplicative group of the field [Formula: see text] on the set of Lee-compositions. We show some useful properties of certain sums of Lee-numbers, which are the eigenvalues of the Lee association scheme, appearing in the linear programming problem of linear Lee codes. Using the additional equality constraints, we formulate the linear programming problem of linear Lee codes in a very compact form, leading to a fast execution, which allows to efficiently compute the bounds for large parameter values of the linear codes.

  7. Introduction to linear elasticity

    CERN Document Server

    Gould, Phillip L

    2013-01-01

    Introduction to Linear Elasticity, 3rd Edition, provides an applications-oriented grounding in the tensor-based theory of elasticity for students in mechanical, civil, aeronautical, and biomedical engineering, as well as materials and earth science. The book is distinct from the traditional text aimed at graduate students in solid mechanics by introducing the subject at a level appropriate for advanced undergraduate and beginning graduate students. The author's presentation allows students to apply the basic notions of stress analysis and move on to advanced work in continuum mechanics, plasticity, plate and shell theory, composite materials, viscoelasticity and finite method analysis. This book also:  Emphasizes tensor-based approach while still distilling down to explicit notation Provides introduction to theory of plates, theory of shells, wave propagation, viscoelasticity and plasticity accessible to advanced undergraduate students Appropriate for courses following emerging trend of teaching solid mechan...

  8. Linear step drive

    International Nuclear Information System (INIS)

    Haniger, L.; Elger, R.; Kocandrle, L.; Zdebor, J.

    1986-01-01

    A linear step drive is described developed in Czechoslovak-Soviet cooperation and intended for driving WWER-1000 control rods. The functional principle is explained of the motor and the mechanical and electrical parts of the drive, power control, and the indicator of position are described. The motor has latches situated in the reactor at a distance of 3 m from magnetic armatures, it has a low structural height above the reactor cover, which suggests its suitability for seismic localities. Its magnetic circuits use counterpoles; the mechanical shocks at the completion of each step are damped using special design features. The position indicator is of a special design and evaluates motor position within ±1% of total travel. A drive diagram and the flow chart of both the control electronics and the position indicator are presented. (author) 4 figs

  9. Linear pulse amplifier

    International Nuclear Information System (INIS)

    Tjutju, R.L.

    1977-01-01

    Pulse amplifier is standard significant part of spectrometer. Apart from other type of amplification, it's a combination of amplification and pulse shaping. Because of its special purpose the device should fulfill the following : High resolution is desired to gain a high yield comparable to its actual state of condition. High signal to noise is desired to nhν resolution. High linearity to facilitate calibration. A good overload recovery, in order to the device will capable of analizing a low energy radiation which appear joinly on the high energy fields. Other expections of the device are its economical and practical use its extentive application. For that reason it's built on a standard NIM principle. Taking also into account the above mentioned considerations. High quality component parts are used throughout, while its availability in the domestic market is secured. (author)

  10. Linear Accelerator Laboratory

    International Nuclear Information System (INIS)

    1976-01-01

    This report covers the activity of the Linear Accelerator Laboratory during the period June 1974-June 1976. The activity of the Laboratory is essentially centered on high energy physics. The main activities were: experiments performed with the colliding rings (ACO), construction of the new colliding rings and beginning of the work at higher energy (DCI), bubble chamber experiments with the CERN PS neutrino beam, counter experiments with CERN's PS and setting-up of equipment for new experiments with CERN's SPS. During this period a project has also been prepared for an experiment with the new PETRA colliding ring at Hamburg. On the other hand, intense collaboration with the LURE Laboratory, using the electron synchrotron radiation emitted by ACO and DCI, has been developed [fr

  11. HEAVY ION LINEAR ACCELERATOR

    Science.gov (United States)

    Van Atta, C.M.; Beringer, R.; Smith, L.

    1959-01-01

    A linear accelerator of heavy ions is described. The basic contributions of the invention consist of a method and apparatus for obtaining high energy particles of an element with an increased charge-to-mass ratio. The method comprises the steps of ionizing the atoms of an element, accelerating the resultant ions to an energy substantially equal to one Mev per nucleon, stripping orbital electrons from the accelerated ions by passing the ions through a curtain of elemental vapor disposed transversely of the path of the ions to provide a second charge-to-mass ratio, and finally accelerating the resultant stripped ions to a final energy of at least ten Mev per nucleon.

  12. Linear absorptive dielectrics

    Science.gov (United States)

    Tip, A.

    1998-06-01

    Starting from Maxwell's equations for a linear, nonconducting, absorptive, and dispersive medium, characterized by the constitutive equations D(x,t)=ɛ1(x)E(x,t)+∫t-∞dsχ(x,t-s)E(x,s) and H(x,t)=B(x,t), a unitary time evolution and canonical formalism is obtained. Given the complex, coordinate, and frequency-dependent, electric permeability ɛ(x,ω), no further assumptions are made. The procedure leads to a proper definition of band gaps in the periodic case and a new continuity equation for energy flow. An S-matrix formalism for scattering from lossy objects is presented in full detail. A quantized version of the formalism is derived and applied to the generation of Čerenkov and transition radiation as well as atomic decay. The last case suggests a useful generalization of the density of states to the absorptive situation.

  13. Computer Program For Linear Algebra

    Science.gov (United States)

    Krogh, F. T.; Hanson, R. J.

    1987-01-01

    Collection of routines provided for basic vector operations. Basic Linear Algebra Subprogram (BLAS) library is collection from FORTRAN-callable routines for employing standard techniques to perform basic operations of numerical linear algebra.

  14. Quaternion Linear Canonical Transform Application

    OpenAIRE

    Bahri, Mawardi

    2015-01-01

    Quaternion linear canonical transform (QLCT) is a generalization of the classical linear canonical transfom (LCT) using quaternion algebra. The focus of this paper is to introduce an application of the QLCT to study of generalized swept-frequency filter

  15. Recursive Algorithm For Linear Regression

    Science.gov (United States)

    Varanasi, S. V.

    1988-01-01

    Order of model determined easily. Linear-regression algorithhm includes recursive equations for coefficients of model of increased order. Algorithm eliminates duplicative calculations, facilitates search for minimum order of linear-regression model fitting set of data satisfactory.

  16. Dynamical systems and linear algebra

    OpenAIRE

    Colonius, Fritz (Prof.)

    2007-01-01

    Dynamical systems and linear algebra / F. Colonius, W. Kliemann. - In: Handbook of linear algebra / ed. by Leslie Hogben. - Boca Raton : Chapman & Hall/CRC, 2007. - S. 56,1-56,22. - (Discrete mathematics and its applications)

  17. Linear spaces: history and theory

    OpenAIRE

    Albrecht Beutelspracher

    1990-01-01

    Linear spaces belong to the most fundamental geometric and combinatorial structures. In this paper I would like to give an onerview about the theory of embedding finite linear spaces in finite projective planes.

  18. Linear versus non-linear supersymmetry, in general

    Energy Technology Data Exchange (ETDEWEB)

    Ferrara, Sergio [Theoretical Physics Department, CERN,CH-1211 Geneva 23 (Switzerland); INFN - Laboratori Nazionali di Frascati,Via Enrico Fermi 40, I-00044 Frascati (Italy); Department of Physics and Astronomy, UniversityC.L.A.,Los Angeles, CA 90095-1547 (United States); Kallosh, Renata [SITP and Department of Physics, Stanford University,Stanford, California 94305 (United States); Proeyen, Antoine Van [Institute for Theoretical Physics, Katholieke Universiteit Leuven,Celestijnenlaan 200D, B-3001 Leuven (Belgium); Wrase, Timm [Institute for Theoretical Physics, Technische Universität Wien,Wiedner Hauptstr. 8-10, A-1040 Vienna (Austria)

    2016-04-12

    We study superconformal and supergravity models with constrained superfields. The underlying version of such models with all unconstrained superfields and linearly realized supersymmetry is presented here, in addition to the physical multiplets there are Lagrange multiplier (LM) superfields. Once the equations of motion for the LM superfields are solved, some of the physical superfields become constrained. The linear supersymmetry of the original models becomes non-linearly realized, its exact form can be deduced from the original linear supersymmetry. Known examples of constrained superfields are shown to require the following LM’s: chiral superfields, linear superfields, general complex superfields, some of them are multiplets with a spin.

  19. Linear versus non-linear supersymmetry, in general

    International Nuclear Information System (INIS)

    Ferrara, Sergio; Kallosh, Renata; Proeyen, Antoine Van; Wrase, Timm

    2016-01-01

    We study superconformal and supergravity models with constrained superfields. The underlying version of such models with all unconstrained superfields and linearly realized supersymmetry is presented here, in addition to the physical multiplets there are Lagrange multiplier (LM) superfields. Once the equations of motion for the LM superfields are solved, some of the physical superfields become constrained. The linear supersymmetry of the original models becomes non-linearly realized, its exact form can be deduced from the original linear supersymmetry. Known examples of constrained superfields are shown to require the following LM’s: chiral superfields, linear superfields, general complex superfields, some of them are multiplets with a spin.

  20. 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).