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...

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.

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.

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.

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.

Active noise control in a duct to cancel broadband noise

Science.gov (United States)

Chen, Kuan-Chun; Chang, Cheng-Yuan; Kuo, Sen M.

2017-09-01

The paper presents cancelling duct noises by using the active noise control (ANC) techniques. We use the single channel feed forward algorithm with feedback neutralization to realize ANC. Several kinds of ducts noises including tonal noises, sweep tonal signals, and white noise had investigated. Experimental results show that the proposed ANC system can cancel these noises in a PVC duct very well. The noise reduction of white noise can be up to 20 dB.

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.

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.

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...

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.

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...

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

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

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...

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.

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...

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.

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...

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.

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...

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

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....

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...

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....

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

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.

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

NARCIS (Netherlands)

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

2012-01-01

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

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

NARCIS (Netherlands)

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

2012-01-01

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

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

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

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.

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

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.

Analysis, Design, and Evaluation of Acoustic Feedback Cancellation Systems for Hearing Aids

DEFF Research Database (Denmark)

Guo, Meng

2013-01-01

Acoustic feedback problems occur when the output loudspeaker signal of an audio system is partly returned to the input microphone via an acoustic coupling through the air. This problem often causes significant performance degradations in applications such as public address systems and hearing aids....... In the worst case, the audio system becomes unstable and howling occurs. In this work, first we analyze a general multiple microphone audio processing system, where a cancellation system using adaptive filters is used to cancel the effect of acoustic feedback. We introduce and derive an accurate approximation...... in acoustic feedback cancellation for hearing aids. It utilizes a probe noise signal which is generated with a specific characteristic so that it can facilitate an unbiased adaptive filter estimation with fast tracking of feedback path variations/changes despite its low signal level. We show in a hearing aid...

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

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

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.

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.

Design and analysis of a BLPC vocoder-based adaptive feedback cancellation with probe noise

DEFF Research Database (Denmark)

Anand, Ankita; Kar, Asutosh; Swamy, M.N.S.

2017-01-01

a BLPC vocoderbased adaptive feedback canceller with probe noise with an objective of reducing the low-frequency bias in digital hearing-aids. A step-wise mathematical analysis of the proposed feedback canceller is presented employing the recursive least square and normalized least mean square adaptive......The band-limited linear predictive coding (BLPC) vocoder-based adaptive feedback cancellation (AFC) removes the high-frequency bias, while the low frequency bias persists between the desired input signal and the loudspeaker signal in the estimate of the feedback path. In this paper, we present...... algorithms. It is observed that the optimal solution of the feedback path is unbiased for an unshaped probe noise, but is biased for a shaped probe signal; the bias term does not consist of correlation between the desired input and the loudspeaker output. The identifiability conditions are analysed...

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

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.

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...

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 ...

An improved probe noise approach for acoustic feedback cancellation

DEFF Research Database (Denmark)

Guo, Meng; Jensen, Søren Holdt; Jensen, Jesper

2012-01-01

The perhaps most challenging problem in acoustic feedback cancellation using adaptive filters is the bias problem. It is well-known that a probe noise approach can effectively prevent this problem. However, when the probe noise must be inaudible and the steady-state error of the adaptive filter...... expressions for its system behavior, and through a simulation experiment in an acoustic feedback cancellation system....

First Test of Fan Active Noise Control (ANC) Completed

Science.gov (United States)

2005-01-01

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

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,

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.

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

A study on the method for cancelling the background noise of the impact signal

International Nuclear Information System (INIS)

Kim, J. S.; Ham, C. S.; Park, J. H.

1998-01-01

In this paper, we compared the noise canceller (time domain analysis method) to the spectral subtraction (frequency domain analysis method) for cancelling background noise when the Loose Part Monitoring System's accelerometers combined the noise signal with the impact signal if the impact signal exists. In the operation of a nuclear power plant monitoring, alarm triggering occurs due to a peak signal in the background noise, an amplitude increase by component operation such as control rod movement or abrupt pump operation. This operation causes the background noise in LPMS. Thus this noise inputs to LPMS together with the impact signal. In case that this noise amplitude is very large comparing to that of the impact signal, we may not analyze the impact position and mass estimation. We analyzed two methods for cancelling background noise. First, we evaluate the signal to noise ratio utilizing the noise canceller. Second, we evaluate the signal to noise ratio utilizing the spectral subtraction. The evaluation resulted superior the noise canceller to the spectral subtraction on the signal to noise ratio

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)

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.

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

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

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.

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.

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

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.

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

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)

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.

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....

Novel Acoustic Feedback Cancellation Approaches In Hearing Aid Applications Using Probe Noise and Probe Noise Enhancement

DEFF Research Database (Denmark)

Guo, Meng; Jensen, Søren Holdt; Jensen, Jesper

2012-01-01

. In many cases, this bias problem causes the cancellation system to fail. The traditional probe noise approach, where a noise signal is added to the loudspeaker signal can, in theory, prevent the bias. However, in practice, the probe noise level must often be so high that the noise is clearly audible...... and annoying; this makes the traditional probe noise approach less useful in practical applications. In this work, we explain theoretically the decreased convergence rate when using low-level probe noise in the traditional approach, before we propose and study analytically two new probe noise approaches...... the proposed approaches much more attractive in practical applications. We demonstrate this through a simulation experiment with audio signals in a hearing aid acoustic feedback cancellation system, where the convergence rate is improved by as much as a factor of 10....

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.

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

Doubling the spectrum of time-domain induced polarization: removal of non-linear self-potential drift, harmonic noise and spikes, tapered gating, and uncertainty estimation

DEFF Research Database (Denmark)

Olsson, Per-Ivar; Fiandaca, Gianluca; Larsen, Jakob Juul

, a logarithmic gate width distribution for optimizing IP data quality and an estimate of gating uncertainty. Additional steps include modelling and cancelling of non-linear background drift and harmonic noise and a technique for efficiently identifying and removing spikes. The cancelling of non-linear background...... drift is based on a Cole-Cole model which effectively handles current induced electrode polarization drift. The model-based cancelling of harmonic noise reconstructs the harmonic noise as a sum of harmonic signals with a common fundamental frequency. After segmentation of the signal and determining....... The processing steps is successfully applied on full field profile data sets. With the model-based cancelling of harmonic noise, the first usable IP gate is moved one decade closer to time zero. Furthermore, with a Cole-Cole background drift model the shape of the response at late times is accurately retrieved...

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...

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.

Cancelation and its simulation using Matlab according to active noise control case study of automotive noise silencer

Science.gov (United States)

Alfisyahrin; Isranuri, I.

2018-02-01

Active Noise Control is a technique to overcome noisy with noise or sound countered with sound in scientific terminology i.e signal countered with signals. This technique can be used to dampen relevant noise in accordance with the wishes of the engineering task and reducing automotive muffler noise to a minimum. Objective of this study is to develop a Active Noise Control which should cancel the noise of automotive Exhaust (Silencer) through Signal Processing Simulation methods. Noise generator of Active Noise Control is to make the opponent signal amplitude and frequency of the automotive noise. The steps are: Firstly, the noise of automotive silencer was measured to characterize the automotive noise that its amplitude and frequency which intended to be expressed. The opposed sound which having similar character with the signal source should be generated by signal function. A comparison between the data which has been completed with simulation calculations Fourier transform field data is data that has been captured on the muffler (noise silencer) Toyota Kijang Capsule assembly 2009. MATLAB is used to simulate how the signal processing noise generated by exhaust (silencer) using FFT. This opponent is inverted phase signal from the signal source 180° conducted by Instruments of Signal Noise Generators. The process of noise cancelation examined through simulation using computer software simulation. The result is obtained that attenuation of sound (noise cancellation) has a difference of 33.7%. This value is obtained from the comparison of the value of the signal source and the signal value of the opponent. So it can be concluded that the noisy signal can be attenuated by 33.7%.

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...

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...

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…

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,

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

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.

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...

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.

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

Science.gov (United States)

Chen, Yung-Yue

2018-05-08

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

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

Directory of Open Access Journals (Sweden)

Yung-Yue Chen

2018-05-01

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

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)

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.

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.

Investigation of LO-leakage cancellation and DC-offset influence on flicker-noise in X-band mixers

DEFF Research Database (Denmark)

Michaelsen, Rasmus; Johansen, Tom; Tamborg, Kjeld

2012-01-01

This paper describes an investigation on the influences in 1/f noise of LO-leakage and DC-offset cancellation for X-band mixers. Conditions for LO-leakage cancellation and zero DC-offset is derived. Measurements on a double balanced diode mixer shows an improvement in noise figure from 14.3dB to ...

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

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

Active cancellation of probing in linear dipole phased array

CERN Document Server

Singh, Hema; Jha, Rakesh Mohan

2015-01-01

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

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

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.

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)

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

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)

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

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.

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)

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.

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.

Evaluation of the Performance of Feedforward and Recurrent Neural Networks in Active Cancellation of Sound Noise

Directory of Open Access Journals (Sweden)

Mehrshad Salmasi

2012-07-01

Full Text Available Active noise control is based on the destructive interference between the primary noise and generated noise from the secondary source. An antinoise of equal amplitude and opposite phase is generated and combined with the primary noise. In this paper, performance of the neural networks is evaluated in active cancellation of sound noise. For this reason, feedforward and recurrent neural networks are designed and trained. After training, performance of the feedforwrad and recurrent networks in noise attenuation are compared. We use Elman network as a recurrent neural network. For simulations, noise signals from a SPIB database are used. In order to compare the networks appropriately, equal number of layers and neurons are considered for the networks. Moreover, training and test samples are similar. Simulation results show that feedforward and recurrent neural networks present good performance in noise cancellation. As it is seen, the ability of recurrent neural network in noise attenuation is better than feedforward network.

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

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

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.

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...

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

Self-Interference Cancellation Using Time-Domain Phase Noise Estimation in OFDM Full-Duplex Systems

OpenAIRE

Shehata, Heba; Khattab, Tamer

2016-01-01

In full-duplex systems, oscillator phase noise (PN) problem is considered the bottleneck challenge that may face the self-interference cancellation (SIC) stage especially when orthogonal frequency division multiplexing (OFDM) transmission scheme is deployed. Phase noise degrades the SIC performance significantly, if not mitigated before or during the SIC technique. The presence of the oscillator phase noise has different impacts on the transmitted data symbol like common phase error (CPE) and...

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

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%.

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%.

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

Science.gov (United States)

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

2017-12-01

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

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.

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.

Evaluation of the Performance of Feedforward and Recurrent Neural Networks in Active Cancellation of Sound Noise

OpenAIRE

Mehrshad Salmasi; Homayoun Mahdavi-Nasab

2012-01-01

Active noise control is based on the destructive interference between the primary noise and generated noise from the secondary source. An antinoise of equal amplitude and opposite phase is generated and combined with the primary noise. In this paper, performance of the neural networks is evaluated in active cancellation of sound noise. For this reason, feedforward and recurrent neural networks are designed and trained. After training, performance of the feedforwrad and recurrent networks in n...

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)

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)

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.

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.

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.

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

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.

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.

A flicker noise/IM3 cancellation technique for active mixer using negative impedance

NARCIS (Netherlands)

Cheng, W.; Annema, Anne J.; Wienk, Gerhardus J.M.; Nauta, Bram

2013-01-01

Abstract—This paper presents an approach to simultaneously cancel flicker noise and IM3 in Gilbert-type mixers, utilizing negative impedances. For proof of concept, two prototype double-balanced mixers in 0.16- m CMOS are fabricated. The first demonstration mixer chip was optimized for full IM3

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

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.

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.

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...

A Robust Approach For Acoustic Noise Suppression In Speech Using ANFIS

Science.gov (United States)

Martinek, Radek; Kelnar, Michal; Vanus, Jan; Bilik, Petr; Zidek, Jan

2015-11-01

The authors of this article deals with the implementation of a combination of techniques of the fuzzy system and artificial intelligence in the application area of non-linear noise and interference suppression. This structure used is called an Adaptive Neuro Fuzzy Inference System (ANFIS). This system finds practical use mainly in audio telephone (mobile) communication in a noisy environment (transport, production halls, sports matches, etc). Experimental methods based on the two-input adaptive noise cancellation concept was clearly outlined. Within the experiments carried out, the authors created, based on the ANFIS structure, a comprehensive system for adaptive suppression of unwanted background interference that occurs in audio communication and degrades the audio signal. The system designed has been tested on real voice signals. This article presents the investigation and comparison amongst three distinct approaches to noise cancellation in speech; they are LMS (least mean squares) and RLS (recursive least squares) adaptive filtering and ANFIS. A careful review of literatures indicated the importance of non-linear adaptive algorithms over linear ones in noise cancellation. It was concluded that the ANFIS approach had the overall best performance as it efficiently cancelled noise even in highly noise-degraded speech. Results were drawn from the successful experimentation, subjective-based tests were used to analyse their comparative performance while objective tests were used to validate them. Implementation of algorithms was experimentally carried out in Matlab to justify the claims and determine their relative performances.

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.

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.

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.

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...

Possibilities of an efficient noise damping in the large gas-fueled boilers

International Nuclear Information System (INIS)

1994-01-01

This report describes a series of tests with the objective to develop a technique for active control, and suppression of unstable conditions of combustion in gas-fired boiler systems. The primary target group for this report are persons within the energy sector, including especially users of natural gas. Focus has been given more to the practical possibilities and limitations than to complicated acoustical theory. The technical background for this work is constituted by the latest innovations within active noise cancelling methods for low frequent noise in channel- and pipe shaped systems, such as heating-, air-and exhaust systems. The fundamental idea has been to implement already functioning and commercial available systems for active noise cancellation in an modified setup to a gasfired boiler system. A few examples of similar attempts are given. Latest a relatively successful example of actively controlling an unstable combustion in a channel shaped burner system (CNRS Paris 1993). The conclusion of the project is anyway, that unstable combustion in larger gasfired boilers can not be controlled by the active methods and systems known today. The reasons for this are: the flame and thus the noise source are distributed in space, the acoustical wave-propagation in the combustion chamber is complex and with low damping, several loops for unstable combustion are possible. Within this project special sensors for high temperatures are developed, e.g.: probe microphones, high temperature turbulence screens and a system for measuring of the variations in UV-light emission form the flame. These systems are applicable also for more 'traditional active noise cancellation' of low-frequent noise in exhaust pipes and chimneys. (au)

Active Electromagnetic Interference Cancelation for Automotive Switch-Mode Audio Power Amplifiers

DEFF Research Database (Denmark)

Knott, Arnold; Pfaffinger, Gerhard; Andersen, Michael A. E.

2009-01-01

Recent trends in the automotive audio industry have shown the importance of active noise cancelation (ANC) for major improvements in mobile entertainment environments. These approaches target the acoustical noise in the cabin and superimpose an inverse noise signal to cancel disturbances. Electro......Recent trends in the automotive audio industry have shown the importance of active noise cancelation (ANC) for major improvements in mobile entertainment environments. These approaches target the acoustical noise in the cabin and superimpose an inverse noise signal to cancel disturbances...

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?

Correlated Levy Noise in Linear Dynamical Systems

International Nuclear Information System (INIS)

Srokowski, T.

2011-01-01

Linear dynamical systems, driven by a non-white noise which has the Levy distribution, are analysed. Noise is modelled by a specific stochastic process which is defined by the Langevin equation with a linear force and the Levy distributed symmetric white noise. Correlation properties of the process are discussed. The Fokker-Planck equation driven by that noise is solved. Distributions have the Levy shape and their width, for a given time, is smaller than for processes in the white noise limit. Applicability of the adiabatic approximation in the case of the linear force is discussed. (author)

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.

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

Gravitational wave searches with pulsar timing arrays: Cancellation of clock and ephemeris noises

Science.gov (United States)

Tinto, Massimo

2018-04-01

We propose a data processing technique to cancel monopole and dipole noise sources (such as clock and ephemeris noises, respectively) in pulsar timing array searches for gravitational radiation. These noises are the dominant sources of correlated timing fluctuations in the lower-part (≈10-9-10-8 Hz ) of the gravitational wave band accessible by pulsar timing experiments. After deriving the expressions that reconstruct these noises from the timing data, we estimate the gravitational wave sensitivity of our proposed processing technique to single-source signals to be at least one order of magnitude higher than that achievable by directly processing the timing data from an equal-size array. Since arrays can generate pairs of clock and ephemeris-free timing combinations that are no longer affected by correlated noises, we implement with them the cross-correlation statistic to search for an isotropic stochastic gravitational wave background. We find the resulting optimal signal-to-noise ratio to be more than one order of magnitude larger than that obtainable by correlating pairs of timing data from arrays of equal size.

Applications of adaptive filters in active noise control

Science.gov (United States)

Darlington, Paul

The active reduction of acoustic noise is achieved by the addition of a cancelling acoustic signal to the unwanted sound. Successful definition of the cancelling signal amounts to a system identification problem. Recent advances in adaptive signal processing have allowed this problem to be tackled using adaptive filters, which offer significant advantages over conventional solutions. The extension of adaptive noise cancelling techniques, which were developed in the electrical signal conditioning context, to the control of acoustic systems is studied. An analysis is presented of the behavior of the Widrow-Hoff LMS adaptive noise canceller with a linear filter in its control loop. The active control of plane waves propagating axially in a hardwalled duct is used as a motivating model problem. The model problem also motivates the study of the effects of feedback around an LMS adaptive filter. An alternative stochastic gradient algorithm for controlling adaptive filters in the presence of feedback is presented.

A 0.18 μm biosensor front-end based on 1/f noise, distortion cancelation and chopper stabilization techniques.

Science.gov (United States)

Balasubramanian, Viswanathan; Ruedi, Pierre-Francois; Temiz, Yuksel; Ferretti, Anna; Guiducci, Carlotta; Enz

2013-10-01

This paper presents a novel sensor front-end circuit that addresses the issues of 1/f noise and distortion in a unique way by using canceling techniques. The proposed front-end is a fully differential transimpedance amplifier (TIA) targeted for current mode electrochemical biosensing applications. In this paper, we discuss the architecture of this canceling based front-end and the optimization methods followed for achieving low noise, low distortion performance at minimum current consumption are presented. To validate the employed canceling based front-end, it has been realized in a 0.18 μm CMOS process and the characterization results are presented. The front-end has also been tested as part of a complete wireless sensing system and the cyclic voltammetry (CV) test results from electrochemical sensors are provided. Overall current consumption in the front-end is 50 μA while operating on a 1.8 V supply.

Improving the accuracy of smart devices to measure noise exposure.

Science.gov (United States)

Roberts, Benjamin; Kardous, Chucri; Neitzel, Richard

2016-11-01

Occupational noise exposure is one of the most frequent hazards present in the workplace; up to 22 million workers have potentially hazardous noise exposures in the U.S. As a result, noise-induced hearing loss is one of the most common occupational injuries in the U.S. Workers in manufacturing, construction, and the military are at the highest risk for hearing loss. Despite the large number of people exposed to high levels of noise at work, many occupations have not been adequately evaluated for noise exposure. The objective of this experiment was to investigate whether or not iOS smartphones and other smart devices (Apple iPhones and iPods) could be used as reliable instruments to measure noise exposures. For this experiment three different types of microphones were tested with a single model of iPod and three generations of iPhones: the internal microphones on the device, a low-end lapel microphone, and a high-end lapel microphone marketed as being compliant with the International Electrotechnical Commission's (IEC) standard for a Class 2-microphone. All possible combinations of microphones and noise measurement applications were tested in a controlled environment using several different levels of pink noise ranging from 60-100 dBA. Results were compared to simultaneous measurements made using a Type 1 sound level measurement system. Analysis of variance and Tukey's honest significant difference (HSD) test were used to determine if the results differed by microphone or noise measurement application. Levels measured with external microphones combined with certain noise measurement applications did not differ significantly from levels measured with the Type 1 sound measurement system. Results showed that it may be possible to use iOS smartphones and smart devices, with specific combinations of measurement applications and calibrated external microphones, to collect reliable, occupational noise exposure data under certain conditions and within the limitations of the

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.

Improved prediction error filters for adaptive feedback cancellation in hearing aids

DEFF Research Database (Denmark)

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

2013-01-01

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

Fiber Optic Microphone

Science.gov (United States)

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

1999-01-01

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

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

Science.gov (United States)

Yoshinaga, Hiroshi; Mizutani, Koichi; Wakatsuki, Naoto

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

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.

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.

Fully integrated low-noise readout circuit with automatic offset cancellation loop for capacitive microsensors.

Science.gov (United States)

Song, Haryong; Park, Yunjong; Kim, Hyungseup; Cho, Dong-Il Dan; Ko, Hyoungho

2015-10-14

Capacitive sensing schemes are widely used for various microsensors; however, such microsensors suffer from severe parasitic capacitance problems. This paper presents a fully integrated low-noise readout circuit with automatic offset cancellation loop (AOCL) for capacitive microsensors. The output offsets of the capacitive sensing chain due to the parasitic capacitances and process variations are automatically removed using AOCL. The AOCL generates electrically equivalent offset capacitance and enables charge-domain fine calibration using a 10-bit R-2R digital-to-analog converter, charge-transfer switches, and a charge-storing capacitor. The AOCL cancels the unwanted offset by binary-search algorithm based on 10-bit successive approximation register (SAR) logic. The chip is implemented using 0.18 μm complementary metal-oxide-semiconductor (CMOS) process with an active area of 1.76 mm². The power consumption is 220 μW with 3.3 V supply. The input parasitic capacitances within the range of -250 fF to 250 fF can be cancelled out automatically, and the required calibration time is lower than 10 ms.

Fully Integrated Low-Noise Readout Circuit with Automatic Offset Cancellation Loop for Capacitive Microsensors

Directory of Open Access Journals (Sweden)

Haryong Song

2015-10-01

Full Text Available Capacitive sensing schemes are widely used for various microsensors; however, such microsensors suffer from severe parasitic capacitance problems. This paper presents a fully integrated low-noise readout circuit with automatic offset cancellation loop (AOCL for capacitive microsensors. The output offsets of the capacitive sensing chain due to the parasitic capacitances and process variations are automatically removed using AOCL. The AOCL generates electrically equivalent offset capacitance and enables charge-domain fine calibration using a 10-bit R-2R digital-to-analog converter, charge-transfer switches, and a charge-storing capacitor. The AOCL cancels the unwanted offset by binary-search algorithm based on 10-bit successive approximation register (SAR logic. The chip is implemented using 0.18 μm complementary metal-oxide-semiconductor (CMOS process with an active area of 1.76 mm2. The power consumption is 220 μW with 3.3 V supply. The input parasitic capacitances within the range of −250 fF to 250 fF can be cancelled out automatically, and the required calibration time is lower than 10 ms.

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.

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)

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.

Adaptive cancellation of motion artifact in wearable biosensors.

Science.gov (United States)

Yousefi, Rasoul; Nourani, Mehrdad; Panahi, Issa

2012-01-01

The performance of wearable biosensors is highly influenced by motion artifact. In this paper, a model is proposed for analysis of motion artifact in wearable photoplethysmography (PPG) sensors. Using this model, we proposed a robust real-time technique to estimate fundamental frequency and generate a noise reference signal. A Least Mean Square (LMS) adaptive noise canceler is then designed and validated using our synthetic noise generator. The analysis and results on proposed technique for noise cancellation shows promising performance.

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....

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.

Adaptive Feedforward Cancellation of Sinusoidal Disturbances in Superconducting RF Cavities

CERN Document Server

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

2004-01-01

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

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

CERN Document Server

Schobben, Daniel W E

2001-01-01

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

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.)

Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation

Directory of Open Access Journals (Sweden)

Schüldt Christian

2007-01-01

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

Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation

Directory of Open Access Journals (Sweden)

Fredric Lindstrom

2007-02-01

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

Effects of venting on wind noise levels measured at the eardrum.

Science.gov (United States)

Chung, King

2013-01-01

Wind noise can be a nuisance to hearing aid users. With the advent of sophisticated feedback reduction algorithms, people with higher degrees of hearing loss are fit with larger vents than previously allowed, and more people with lesser degrees of hearing loss are fit with open hearing aids. The purpose of this study was to examine the effects of venting on wind noise levels in the ear canal for hearing aids with omnidirectional and directional microphones. Two behind-the-ear hearing aids were programmed when they were worn on a Knowles Electronics Manikin for Acoustic Research. The hearing aid worn on the right ear was programmed to the omnidirectional microphone mode and the one on the left to the directional microphone mode. The hearing aids were adjusted to linear amplification with flat frequency response in an anechoic chamber. Gains below 10 dB were used to avoid output limiting of wind noise levels at low input levels. Wind noise samples were recorded at the eardrum location in a wind tunnel at wind velocities ranging from a gentle to a strong breeze. The hearing aids were coupled to #13 tubings (i.e., open vent), or conventional skeleton earmolds with no vent, pressure vents, or 3mm vents. Polar and spectral characteristics of wind noise were analyzed off-line using MatLab programs. Wind noise levels in the ear canals were mostly predicted by vent-induced frequency response changes in the conventional earmold conditions for both omnidirectional and directional hearing aids. The open vent condition, however, yielded the lowest levels, which could not be entirely predicted by the frequency response changes of the hearing aids. This indicated that a wind-related vent effect permitted an additional amount of sound reduction in the ear canal, which could not be explained by known vent effects. For the microphone location, form factor, and gain settings tested, open fit hearing aids yielded lower noise levels at the eardrum location than conventional behind

Validation of an Aero-Acoustic Wind Turbine Noise Model Using Advanced Noise Source Measurements of a 500kW Turbine

DEFF Research Database (Denmark)

Bertagnolio, Franck; Aagaard Madsen, Helge; Fischer, Andreas

2016-01-01

rotor noise model is presented. It includes the main sources of aeroacoustic noise from wind turbines: turbulent inflow, trailing edge and stall noise. The noise measured by one microphone located directly downstream of the wind turbine is compared to the model predictions at the microphone location....... A good qualitative agreement is found. When wind speed increases, the rotor noise model shows that at high frequencies the stall noise becomes dominant. It also shows that turbulent inflow noise is dominant at low frequencies for all wind speeds and that trailing edge noise is dominant at low wind speeds...

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

On The Use of A Phase Modulation Method for Decorrelation in Acoustic Feedback Cancellation

DEFF Research Database (Denmark)

Guo, Meng; Jensen, Søren Holdt; Jensen, Jesper

2012-01-01

A major problem in using an adaptive filter in acoustic feedback cancellation systems is that the loudspeaker signal is correlated with the signals entering the microphones of the audio system, leading to biased filter estimates. One possible solution for reducing this problem is by means...... of decorrelation. In this work, we study a subband phase modulation method, which was originally proposed for decorrelation in multichannel acoustic echo cancellation systems. We determine if this method is effective for decorrelation in acoustic feedback cancellation systems by comparing it to a structurally...... similar frequency shifting decorrelation method. We show that the phase modulation method is suitable for decorrelation in a hearing aid acoustic feedback cancellation system, although the frequency shifting method is in general slightly more effective....

Chip-size-packaged silicon microphones [for hearing instruments

DEFF Research Database (Denmark)

Müllenborn, Matthias; Rombach, Pirmin; Klein, Udo

2001-01-01

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

A 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

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.

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.)

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.

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.

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...

Research status on aero-acoustic noise from wind turbine blades

International Nuclear Information System (INIS)

Yang, B

2013-01-01

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

Seismic Linear Noise Attenuation with Use of Radial Transform

Science.gov (United States)

Szymańska-Małysa, Żaneta

2018-03-01

One of the goals of seismic data processing is to attenuate the recorded noise in order to enable correct interpretation of the image. Radial transform has been used as a very effective tool in the attenuation of various types of linear noise, both numerical and real (such as ground roll, direct waves, head waves, guided waves etc). The result of transformation from offset - time (X - T) domain into apparent velocity - time (R - T) domain is frequency separation between reflections and linear events. In this article synthetic and real seismic shot gathers were examined. One example was targeted at far offset area of dataset where reflections and noise had similar apparent velocities and frequency bands. Another example was a result of elastic modelling where linear artefacts were produced. Bandpass filtering and scaling operation executed in radial domain attenuated all discussed types of linear noise very effectively. After noise reduction all further processing steps reveal better results, especially velocity analysis, migration and stacking. In all presented cases signal-to-noise ratio was significantly increased and reflections covered previously by noise were revealed. Power spectra of filtered seismic records preserved real dynamics of reflections.

Interpixel crosstalk cancellation on holographic memory

Science.gov (United States)

Ishii, Toshiki; Fujimura, Ryushi

2017-09-01

In holographic memory systems, there have been no practical techniques to minimize interpixel crosstalk thus far. We developed an interpixel crosstalk cancellation technique using a checkerboard phase pattern with a phase difference of π/2, which can decrease the size of the spatial filter along the Fourier plane with the signal-to-noise ratio (SNR) kept high. This interpixel crosstalk cancellation technique is simple because it requires only one phase plate in the signal beam path. We verified the effect of such a cancellation technique by simulation. The improvement of SNR is maximized to 6.5 dB when the filter size specified in the Nyquist areal ratio is approximately 1.05 in ideal optical systems with no other fixed noise. The proposed technique can improve SNR by 0.85 in an assumed monocular architecture at an actual noise intensity. This improvement of SNR is very useful for realizing high-density recording or enhancing system robustness.

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)

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.

Towards a better understanding of helicopter external noise

Science.gov (United States)

Damongeot, A.; Dambra, F.; Masure, B.

The problem of helicopter external noise generation is studied taking into consideration simultaneously the multiple noise sources: rotor rotational-, rotor broadband -, and engine noise. The main data are obtained during flight tests of the rather quiet AS 332 Super Puma. The flight procedures settled by ICAO for noise regulations are used: horizontal flyover at 90 percent of the maximum speed, approach at minimum power velocity, take-off at best rate of climb. Noise source levels are assessed through narrow band analysis of ground microphone recordings, ground measurements of engine noise and theoretical means. With the perceived noise level unit used throughout the study, relative magnitude of noise sources is shown to be different from that obtained with linear noise unit. A parametric study of the influence of some helicopter parameters on external noise has shown that thickness-tapered, chord-tapered, and swept-back blade tips are good means to reduce the overall noise level in flyover and approach.

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.

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

Active3 noise reduction

International Nuclear Information System (INIS)

Holzfuss, J.

1996-01-01

Noise reduction is a problem being encountered in a variety of applications, such as environmental noise cancellation, signal recovery and separation. Passive noise reduction is done with the help of absorbers. Active noise reduction includes the transmission of phase inverted signals for the cancellation. This paper is about a threefold active approach to noise reduction. It includes the separation of a combined source, which consists of both a noise and a signal part. With the help of interaction with the source by scanning it and recording its response, modeling as a nonlinear dynamical system is achieved. The analysis includes phase space analysis and global radial basis functions as tools for the prediction used in a subsequent cancellation procedure. Examples are given which include noise reduction of speech. copyright 1996 American Institute of Physics

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

Background noise levels measured in the NASA Lewis 9- by 15-foot low-speed wind tunnel

Science.gov (United States)

Woodward, Richard P.; Dittmar, James H.; Hall, David G.; Kee-Bowling, Bonnie

1994-01-01

The acoustic capability of the NASA Lewis 9 by 15 Foot Low Speed Wind Tunnel has been significantly improved by reducing the background noise levels measured by in-flow microphones. This was accomplished by incorporating streamlined microphone holders having a profile developed by researchers at the NASA Ames Research Center. These new holders were fabricated for fixed mounting on the tunnel wall and for an axially traversing microphone probe which was mounted to the tunnel floor. Measured in-flow noise levels in the tunnel test section were reduced by about 10 dB with the new microphone holders compared with those measured with the older, less refined microphone holders. Wake interference patterns between fixed wall microphones were measured and resulted in preferred placement patterns for these microphones to minimize these effects. Acoustic data from a model turbofan operating in the tunnel test section showed that results for the fixed and translating microphones were equivalent for common azimuthal angles, suggesting that the translating microphone probe, with its significantly greater angular resolution, is preferred for sideline noise measurements. Fixed microphones can provide a local check on the traversing microphone data quality, and record acoustic performance at other azimuthal angles.

Non-linearity consideration when analyzing reactor noise statistical characteristics. [BWR

Energy Technology Data Exchange (ETDEWEB)

Kebadze, B V; Adamovski, L A

1975-06-01

Statistical characteristics of boiling water reactor noise in the vicinity of stability threshold are studied. The reactor is considered as a non-linear system affected by random perturbations. To solve a non-linear problem the principle of statistical linearization is used. It is shown that the halfwidth of resonance peak in neutron power noise spectrum density as well as the reciprocal of noise dispersion, which are used in predicting a stable operation theshold, are different from zero both within and beyond the stability boundary the determination of which was based on linear criteria.

Regression of non-linear coupling of noise in LIGO detectors

Science.gov (United States)

Da Silva Costa, C. F.; Billman, C.; Effler, A.; Klimenko, S.; Cheng, H.-P.

2018-03-01

In 2015, after their upgrade, the advanced Laser Interferometer Gravitational-Wave Observatory (LIGO) detectors started acquiring data. The effort to improve their sensitivity has never stopped since then. The goal to achieve design sensitivity is challenging. Environmental and instrumental noise couple to the detector output with different, linear and non-linear, coupling mechanisms. The noise regression method we use is based on the Wiener–Kolmogorov filter, which uses witness channels to make noise predictions. We present here how this method helped to determine complex non-linear noise couplings in the output mode cleaner and in the mirror suspension system of the LIGO detector.

Airframe Noise from a Hybrid Wing Body Aircraft Configuration

Science.gov (United States)

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

2016-01-01

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

Multi-compartment linear noise approximation

International Nuclear Information System (INIS)

Challenger, Joseph D; McKane, Alan J; Pahle, Jürgen

2012-01-01

The ability to quantify the stochastic fluctuations present in biochemical and other systems is becoming increasing important. Analytical descriptions of these fluctuations are attractive, as stochastic simulations are computationally expensive. Building on previous work, a linear noise approximation is developed for biochemical models with many compartments, for example cells. The procedure is then implemented in the software package COPASI. This technique is illustrated with two simple examples and is then applied to a more realistic biochemical model. Expressions for the noise, given in the form of covariance matrices, are presented. (paper)

Removing Background Noise with Phased Array Signal Processing

Science.gov (United States)

Podboy, Gary; Stephens, David

2015-01-01

Preliminary results are presented from a test conducted to determine how well microphone phased array processing software could pull an acoustic signal out of background noise. The array consisted of 24 microphones in an aerodynamic fairing designed to be mounted in-flow. The processing was conducted using Functional Beam forming software developed by Optinav combined with cross spectral matrix subtraction. The test was conducted in the free-jet of the Nozzle Acoustic Test Rig at NASA GRC. The background noise was produced by the interaction of the free-jet flow with the solid surfaces in the flow. The acoustic signals were produced by acoustic drivers. The results show that the phased array processing was able to pull the acoustic signal out of the background noise provided the signal was no more than 20 dB below the background noise level measured using a conventional single microphone equipped with an aerodynamic forebody.

Noise limitations in optical linear algebra processors.

Science.gov (United States)

Batsell, S G; Jong, T L; Walkup, J F; Krile, T F

1990-05-10

A general statistical noise model is presented for optical linear algebra processors. A statistical analysis which includes device noise, the multiplication process, and the addition operation is undertaken. We focus on those processes which are architecturally independent. Finally, experimental results which verify the analytical predictions are also presented.

Efficient multichannel acoustic echo cancellation using constrained tap selection schemes in the subband domain

Science.gov (United States)

Desiraju, Naveen Kumar; Doclo, Simon; Wolff, Tobias

2017-12-01

Acoustic echo cancellation (AEC) is a key speech enhancement technology in speech communication and voice-enabled devices. AEC systems employ adaptive filters to estimate the acoustic echo paths between the loudspeakers and the microphone(s). In applications involving surround sound, the computational complexity of an AEC system may become demanding due to the multiple loudspeaker channels and the necessity of using long filters in reverberant environments. In order to reduce the computational complexity, the approach of partially updating the AEC filters is considered in this paper. In particular, we investigate tap selection schemes which exploit the sparsity present in the loudspeaker channels for partially updating subband AEC filters. The potential for exploiting signal sparsity across three dimensions, namely time, frequency, and channels, is analyzed. A thorough analysis of different state-of-the-art tap selection schemes is performed and insights about their limitations are gained. A novel tap selection scheme is proposed which overcomes these limitations by exploiting signal sparsity while not ignoring any filters for update in the different subbands and channels. Extensive simulation results using both artificial as well as real-world multichannel signals show that the proposed tap selection scheme outperforms state-of-the-art tap selection schemes in terms of echo cancellation performance. In addition, it yields almost identical echo cancellation performance as compared to updating all filter taps at a significantly reduced computational cost.

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)

Evaluation of State-of-the-Art Acoustic Feedback Cancellation Systems for Hearing Aids

DEFF Research Database (Denmark)

Guo, Meng; Jensen, Søren Holdt; Jensen, Jesper

2013-01-01

This research evaluates four state-of-the-art acoustic feedback cancellation systems in hearing aids in terms of the cancellation performance, sound quality degradation, and computational complexity. The authors compared a traditional full-band system to a system with a prediction error method...... in a full band, a subband system, a subband system with frequency shifting, and a recently proposed subband system with a novel probe noise deployment. All systems outperformed the traditional full-band system in cancellation performance, especially the subband system with probe noise is most effective...... for cancellation. However, in all cases there was a trade-off between performance and computational cost. With a 3-times increase in computation load, the probe noise based cancellation system can be realized that functions even in the most challenging feedback situation....

Wind Noise Reduction in a Non-Porous Subsurface Windscreen

Science.gov (United States)

Zuckerwar, Allan J.; Shams, Qamar A.; Knight, H. Keith

2012-01-01

Measurements of wind noise reduction were conducted on a box-shaped, subsurface windscreen made of closed cell polyurethane foam. The windscreen was installed in the ground with the lid flush with the ground surface. The wind was generated by means of a fan, situated on the ground, and the wind speed was measured at the center of the windscreen lid with an ultrasonic anemometer. The wind speed was controlled by moving the fan to selected distances from the windscreen. The wind noise was measured on a PCB Piezotronics 3†electret microphone. Wind noise spectra were measured with the microphone exposed directly to the wind (atop the windscreen lid) and with the microphone installed inside the windscreen. The difference between the two spectra comprises the wind noise reduction. At wind speeds of 3, 5, and 7 m/s, the wind noise reduction is typically 15 dB over the frequency range of 0.1-20 Hz.

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…

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.

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

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

Advanced Noise Control Fan: A 20-Year Retrospective

Science.gov (United States)

Sutliff, Dan

2016-01-01

The ANCF test bed is used for evaluating fan noise reduction concepts, developing noise measurement technologies, and providing a database for Aero-acoustic code development. Rig Capabilities: 4 foot 16 bladed rotor @ 2500 rpm, Auxiliary air delivery system (3 lbm/sec @ 6/12 psi), Variable configuration (rotor pitch angle, stator count/position, duct length), synthetic acoustic noise generation (tone/broadband). Measurement Capabilities: 112 channels dynamic data system, Unique rotating rake mode measuremen, Farfield (variable radius), Duct wall microphones, Stator vane microphones, Two component CTA w/ traversing, ESP for static pressures.

Intensity noise cancellation in solid-state laser at 1.5  μm using SHG depletion as a buffer reservoir.

Science.gov (United States)

Audo, Kevin; Alouini, Mehdi

2018-03-01

An absorption mechanism based on second-harmonic generation (SHG) is successfully implemented as a buffer reservoir in a solid-state Er,Yb:Glass laser emitting at the telecom wavelength. We show that a slight absorption mechanism based on SHG rate conversion of 0.016% using a beta barium borate crystal enables the canceling out of the excess intensity noise at the relaxation oscillation frequency, i.e., 35 dB reduction, as well as canceling the amplified spontaneous emission beating at the free spectral range resonances of the laser lying in the gigahertz range. Laser robustness is discussed.

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

Mitigation of MIMO Co-Channel Interference using robust interference cancellation receiver

DEFF Research Database (Denmark)

Rahman, Muhammad Imadur; De Carvalho, Elisabeth; Prasad, Ramjee

2007-01-01

(STBC) link may become equivalent to an interfering Spatial Multiplexing (SM) link. Using this knowledge and understanding, we propose an interference cancellation receiver robust to different types of MIMO interferers at cell edge for the Downlink (DL) of cellular systems. The receiver systematically...... performs a multiple symbol processing: this is the appropriate processing when the signal of interest or the signal of interferer is correlated across symbols, which is the case for STBC transmission. We evaluated different link combinations in terms of Signal to Interference and Noise Ratio (SINR......) statistics and Bit Error Rate (BER) performance in cellular Orthogonal Frequency Division Multiple Access (OFDMA) systems. We have found that the proposed multiple-symbol linear interference cancellation receiver performs satisfactorily when any kind of single 'logical' stream MIMO scheme is present...

Wiener variable step size and gradient spectral variance smoothing for double-talk-robust acoustic echo cancellation and acoustic feedback cancellation

DEFF Research Database (Denmark)

Gil-Cacho, Jose M.; Van Waterschoot, Toon; Moonen, Marc

2014-01-01

Double-talk (DT)-robust acoustic echo cancellation (AEC) and acoustic feedback cancellation (AFC) are needed in speech communication systems, e.g., in hands-free communication systems and hearing aids. In this paper, we derive a practical and computationally efficient algorithm based...... model and in colored non-stationary noise....

Speech Enhancement by Multichannel Crosstalk Resistant ANC and Improved Spectrum Subtraction

Directory of Open Access Journals (Sweden)

Zeng Qingning

2006-01-01

Full Text Available A scheme combining multichannel crosstalk resistant adaptive noise cancellation (MCRANC algorithm and improved spectrum subtraction (ISS algorithm is presented to enhance noise carrying speech signals. The scheme would permit locating the microphones in close proximity by virtue of using MCRANC which has the capability of removing the crosstalk effect. MCRANC would also permit canceling out nonstationary noise and making the residual noise more stationary for further treatment by ISS algorithm. Experimental results have indicated that this scheme outperforms many commonly used techniques in the sense of SNR improvement and music effect reduction which is an inevitable byproduct of the spectrum subtraction algorithm.

Crosstalk cancellation on linearly and circularly polarized communications satellite links

Science.gov (United States)

Overstreet, W. P.; Bostian, C. W.

1979-01-01

The paper discusses the cancellation network approach for reducing crosstalk caused by depolarization on a dual-polarized communications satellite link. If the characteristics of rain depolarization are sufficiently well known, the cancellation network can be designed in a way that reduces system complexity, the most important parameter being the phase of the cross-polarized signal. Relevant theoretical calculations and experimental data are presented. The simplicity of the cancellation system proposed makes it ideal for use with small domestic or private earth terminals.

Compressed Sensing with Linear Correlation Between Signal and Measurement Noise

DEFF Research Database (Denmark)

Arildsen, Thomas; Larsen, Torben

2014-01-01

reconstruction algorithms, but is not known in existing literature. The proposed technique reduces reconstruction error considerably in the case of linearly correlated measurements and noise. Numerical experiments confirm the efficacy of the technique. The technique is demonstrated with application to low......Existing convex relaxation-based approaches to reconstruction in compressed sensing assume that noise in the measurements is independent of the signal of interest. We consider the case of noise being linearly correlated with the signal and introduce a simple technique for improving compressed...... sensing reconstruction from such measurements. The technique is based on a linear model of the correlation of additive noise with the signal. The modification of the reconstruction algorithm based on this model is very simple and has negligible additional computational cost compared to standard...

A Microwave Photonic Interference Canceller: Architectures, Systems, and Integration

Science.gov (United States)

Chang, Matthew P.

This thesis is a comprehensive portfolio of work on a Microwave Photonic Self-Interference Canceller (MPC), a specialized optical system designed to eliminate interference from radio-frequency (RF) receivers. The novelty and value of the microwave photonic system lies in its ability to operate over bandwidths and frequencies that are orders of magnitude larger than what is possible using existing RF technology. The work begins, in 2012, with a discrete fiber-optic microwave photonic canceller, which prior work had demonstrated as a proof-of-concept, and culminates, in 2017, with the first ever monolithically integrated microwave photonic canceller. With an eye towards practical implementation, the thesis establishes novelty through three major project thrusts. (Fig. 1): (1) Extensive RF and system analysis to develop a full understanding of how, and through what mechanisms, MPCs affect an RF receiver. The first investigations of how a microwave photonic canceller performs in an actual wireless environment and a digital radio are also presented. (2) New architectures to improve the performance and functionality of MPCs, based on the analysis performed in Thrust 1. A novel balanced microwave photonic canceller architecture is developed and experimentally demonstrated. The balanced architecture shows significant improvements in link gain, noise figure, and dynamic range. Its main advantage is its ability to suppress common-mode noise and reduce noise figure by increasing the optical power. (3) Monolithic integration of the microwave photonic canceller into a photonic integrated circuit. This thrust presents the progression of integrating individual discrete devices into their semiconductor equivalent, as well as a full functional and RF analysis of the first ever integrated microwave photonic canceller.

Noise and non-linearities in high-throughput data

International Nuclear Information System (INIS)

Nguyen, Viet-Anh; Lió, Pietro; Koukolíková-Nicola, Zdena; Bagnoli, Franco

2009-01-01

High-throughput data analyses are becoming common in biology, communications, economics and sociology. The vast amounts of data are usually represented in the form of matrices and can be considered as knowledge networks. Spectra-based approaches have proved useful in extracting hidden information within such networks and for estimating missing data, but these methods are based essentially on linear assumptions. The physical models of matching, when applicable, often suggest non-linear mechanisms, that may sometimes be identified as noise. The use of non-linear models in data analysis, however, may require the introduction of many parameters, which lowers the statistical weight of the model. According to the quality of data, a simpler linear analysis may be more convenient than more complex approaches. In this paper, we show how a simple non-parametric Bayesian model may be used to explore the role of non-linearities and noise in synthetic and experimental data sets

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.

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

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

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...

Linear signal noise summer accurately determines and controls S/N ratio

Science.gov (United States)

Sundry, J. L.

1966-01-01

Linear signal noise summer precisely controls the relative power levels of signal and noise, and mixes them linearly in accurately known ratios. The S/N ratio accuracy and stability are greatly improved by this technique and are attained simultaneously.

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

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.

A Two-Sensor Noise Reduction System: Applications for Hands-Free Car Kit

Directory of Open Access Journals (Sweden)

Guérin Alexandre

2003-01-01

Full Text Available This paper presents a two-microphone speech enhancer designed to remove noise in hands-free car kits. The algorithm, based on the magnitude squared coherence, uses speech correlation and noise decorrelation to separate speech from noise. The remaining correlated noise is reduced using cross-spectral subtraction. Particular attention is focused on the estimation of the different spectral densities (noise and noisy signals power spectral densities which are critical for the quality of the algorithm. We also propose a continuous noise estimation, avoiding the need of vocal activity detector. Results on recorded signals are provided, showing the superiority of the two-sensor approach to single microphone techniques.

Active Noise Control Experiments using Sound Energy Flu

Science.gov (United States)

Krause, Uli

2015-03-01

This paper reports on the latest results concerning the active noise control approach using net flow of acoustic energy. The test set-up consists of two loudspeakers simulating the engine noise and two smaller loudspeakers which belong to the active noise system. The system is completed by two acceleration sensors and one microphone per loudspeaker. The microphones are located in the near sound field of the loudspeakers. The control algorithm including the update equation of the feed-forward controller is introduced. Numerical simulations are performed with a comparison to a state of the art method minimising the radiated sound power. The proposed approach is experimentally validated.

Adaptive Nonlinear RF Cancellation for Improved Isolation in Simultaneous Transmit–Receive Systems

Science.gov (United States)

Kiayani, Adnan; Waheed, Muhammad Zeeshan; Anttila, Lauri; Abdelaziz, Mahmoud; Korpi, Dani; Syrjala, Ville; Kosunen, Marko; Stadius, Kari; Ryynanen, Jussi; Valkama, Mikko

2018-05-01

This paper proposes an active radio frequency (RF) cancellation solution to suppress the transmitter (TX) passband leakage signal in radio transceivers supporting simultaneous transmission and reception. The proposed technique is based on creating an opposite-phase baseband equivalent replica of the TX leakage signal in the transceiver digital front-end through adaptive nonlinear filtering of the known transmit data, to facilitate highly accurate cancellation under a nonlinear TX power amplifier (PA). The active RF cancellation is then accomplished by employing an auxiliary transmitter chain, to generate the actual RF cancellation signal, and combining it with the received signal at the receiver (RX) low noise amplifier (LNA) input. A closed-loop parameter learning approach, based on the decorrelation principle, is also developed to efficiently estimate the coefficients of the nonlinear cancellation filter in the presence of a nonlinear TX PA with memory, finite passive isolation, and a nonlinear RX LNA. The performance of the proposed cancellation technique is evaluated through comprehensive RF measurements adopting commercial LTE-Advanced transceiver hardware components. The results show that the proposed technique can provide an additional suppression of up to 54 dB for the TX passband leakage signal at the RX LNA input, even at considerably high transmit power levels and with wide transmission bandwidths. Such novel cancellation solution can therefore substantially improve the TX-RX isolation, hence reducing the requirements on passive isolation and RF component linearity, as well as increasing the efficiency and flexibility of the RF spectrum use in the emerging 5G radio networks.

Active self-testing noise measurement sensors for large-scale environmental sensor networks.

Science.gov (United States)

Domínguez, Federico; Cuong, Nguyen The; Reinoso, Felipe; Touhafi, Abdellah; Steenhaut, Kris

2013-12-13

Large-scale noise pollution sensor networks consist of hundreds of spatially distributed microphones that measure environmental noise. These networks provide historical and real-time environmental data to citizens and decision makers and are therefore a key technology to steer environmental policy. However, the high cost of certified environmental microphone sensors render large-scale environmental networks prohibitively expensive. Several environmental network projects have started using off-the-shelf low-cost microphone sensors to reduce their costs, but these sensors have higher failure rates and produce lower quality data. To offset this disadvantage, we developed a low-cost noise sensor that actively checks its condition and indirectly the integrity of the data it produces. The main design concept is to embed a 13 mm speaker in the noise sensor casing and, by regularly scheduling a frequency sweep, estimate the evolution of the microphone's frequency response over time. This paper presents our noise sensor's hardware and software design together with the results of a test deployment in a large-scale environmental network in Belgium. Our middle-range-value sensor (around €50) effectively detected all experienced malfunctions, in laboratory tests and outdoor deployments, with a few false positives. Future improvements could further lower the cost of our sensor below €10.

Optimization of actuator arrays for aircraft interior noise control

Science.gov (United States)

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

1993-01-01

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

Analysis of Beamformer Directed Single-Channel Noise Reduction System for Hearing Aid Applications

DEFF Research Database (Denmark)

Jensen, Jesper; Pedersen, Michael Syskind

2015-01-01

We study multi-microphone noise reduction systems consisting of a beamformer and a single-channel (SC) noise reduction stage. In particular, we present and analyse a maximum likelihood (ML) method for jointly estimating the target and noise power spectral densities (psd's) entering the SC filter....... We show that the estimators are minimum variance and unbiased, and provide closed-form expressions for their mean-square error (MSE). Furthermore, we show that the MSE of the noise psd estimator is particularly simple: it is independent of target signal characteristics, frequency, and microphone...

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.

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.

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

Science.gov (United States)

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

2017-07-01

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

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.

Field-Deployable Acoustic Digital Systems for Noise Measurement

Science.gov (United States)

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

2000-01-01

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

Visuo-manual tracking: does intermittent control with aperiodic sampling explain linear power and non-linear remnant without sensorimotor noise?

Science.gov (United States)

Gollee, Henrik; Gawthrop, Peter J; Lakie, Martin; Loram, Ian D

2017-11-01

A human controlling an external system is described most easily and conventionally as linearly and continuously translating sensory input to motor output, with the inevitable output remnant, non-linearly related to the input, attributed to sensorimotor noise. Recent experiments show sustained manual tracking involves repeated refractoriness (insensitivity to sensory information for a certain duration), with the temporary 200-500 ms periods of irresponsiveness to sensory input making the control process intrinsically non-linear. This evidence calls for re-examination of the extent to which random sensorimotor noise is required to explain the non-linear remnant. This investigation of manual tracking shows how the full motor output (linear component and remnant) can be explained mechanistically by aperiodic sampling triggered by prediction error thresholds. Whereas broadband physiological noise is general to all processes, aperiodic sampling is associated with sensorimotor decision making within specific frontal, striatal and parietal networks; we conclude that manual tracking utilises such slow serial decision making pathways up to several times per second. The human operator is described adequately by linear translation of sensory input to motor output. Motor output also always includes a non-linear remnant resulting from random sensorimotor noise from multiple sources, and non-linear input transformations, for example thresholds or refractory periods. Recent evidence showed that manual tracking incurs substantial, serial, refractoriness (insensitivity to sensory information of 350 and 550 ms for 1st and 2nd order systems respectively). Our two questions are: (i) What are the comparative merits of explaining the non-linear remnant using noise or non-linear transformations? (ii) Can non-linear transformations represent serial motor decision making within the sensorimotor feedback loop intrinsic to tracking? Twelve participants (instructed to act in three prescribed

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.

Two-wire Interface for Digital Microphones

NARCIS (Netherlands)

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

2003-01-01

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

Two-Wire interface for digital microphones

NARCIS (Netherlands)

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

2005-01-01

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

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)

Optical linear algebra processors - Noise and error-source modeling

Science.gov (United States)

Casasent, D.; Ghosh, A.

1985-01-01

The modeling of system and component noise and error sources in optical linear algebra processors (OLAPs) are considered, with attention to the frequency-multiplexed OLAP. General expressions are obtained for the output produced as a function of various component errors and noise. A digital simulator for this model is discussed.

Optical linear algebra processors: noise and error-source modeling.

Science.gov (United States)

Casasent, D; Ghosh, A

1985-06-01

The modeling of system and component noise and error sources in optical linear algebra processors (OLAP's) are considered, with attention to the frequency-multiplexed OLAP. General expressions are obtained for the output produced as a function of various component errors and noise. A digital simulator for this model is discussed.

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...

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.

Optimization of the GOES-1 Imagers radiometric accuracy: Drift and 1/f noise suppression

International Nuclear Information System (INIS)

Bremer, J.C.; Comeyne, G.J. III

1994-01-01

The raw output of many scanning radiometers is a small, rapidly varying signal superimposed on a large background that varies more slowly, due to thermal drifts and 1/f noise. To isolate the signal, it is necessary to perform a differential measurement: measure a known reference and subtract it from each of the raw outputs, canceling the common-mode background. Calibration is also a differential measurement: the difference between two outputs is divided by the difference between the two known references that produced them to determine the gain. The GOES-I Imager views space as its background subtraction reference and a full-aperture blackbody as its second reference for calibration. The background suppression efficiency of a differential measurement algorithm depends on its timing. The Imager measures space references before and after each scan line and performs interpolated background subtraction: a unique, linearly weighted average of the two references is subtracted from each scene sample in that line, canceling both constant bias and linear drift. One model quantifies the Gaussian noise and 1/f noise terms in the noise equivalent bandwidth, which is minimized to optimize the algorithm. The authors have obtained excellent agreement between the analytical predictions and Monte Carlo computer simulations

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.

Sources and levels of background noise in the NASA Ames 40- by 80-foot wind tunnel

Science.gov (United States)

Soderman, Paul T.

1988-01-01

Background noise levels are measured in the NASA Ames Research Center 40- by 80-Foot Wind Tunnel following installation of a sound-absorbent lining on the test-section walls. Results show that the fan-drive noise dominated the empty test-section background noise at airspeeds below 120 knots. Above 120 knots, the test-section broadband background noise was dominated by wind-induced dipole noise (except at lower harmonics of fan blade-passage tones) most likely generated at the microphone or microphone support strut. Third-octave band and narrow-band spectra are presented for several fan operating conditions and test-section airspeeds. The background noise levels can be reduced by making improvements to the microphone wind screen or support strut. Empirical equations are presented relating variations of fan noise with fan speed or blade-pitch angle. An empirical expression for typical fan noise spectra is also presented. Fan motor electric power consumption is related to the noise generation. Preliminary measurements of sound absorption by the test-section lining indicate that the 152 mm thick lining will adequately absorb test-section model noise at frequencies above 300 Hz.

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

International Nuclear Information System (INIS)

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

2017-01-01

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

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

Science.gov (United States)

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

2017-10-01

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

On spatial spillover in feedforward and feedback noise control

Science.gov (United States)

Xie, Antai; Bernstein, Dennis

2017-03-01

Active feedback noise control for rejecting broadband disturbances must contend with the Bode integral constraint, which implies that suppression over some frequency range gives rise to amplification over another range at the performance microphone. This is called spectral spillover. The present paper deals with spatial spillover, which refers to the amplification of noise at locations where no microphone is located. A spatial spillover function is defined, which is valid for both feedforward and feedback control with scalar and vector control inputs. This function is numerically analyzed and measured experimentally. Obstructions are introduced in the acoustic space to investigate their effect on spatial spillover.

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

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.

Displacement-noise-free gravitational-wave detection with a single Fabry-Perot cavity: A toy model

International Nuclear Information System (INIS)

Tarabrin, Sergey P.; Vyatchanin, Sergey P.

2008-01-01

We propose a detuned Fabry-Perot cavity, pumped through both the mirrors, as a toy model of the gravitational-wave (GW) detector partially free from displacement noise of the test masses. It is demonstrated that the noise of cavity mirrors can be eliminated, but the one of lasers and detectors cannot. The isolation of the GW signal from displacement noise of the mirrors is achieved in a proper linear combination of the cavity output signals. The construction of such a linear combination is possible due to the difference between the reflected and transmitted output signals of detuned cavity. We demonstrate that in low-frequency region the obtained displacement-noise-free response signal is much stronger than the f gw 3 -limited sensitivity of displacement-noise-free interferometers recently proposed by S. Kawamura and Y. Chen. However, the loss of the resonant gain in the noise cancelation procedure results is the sensitivity limitation of our toy model by displacement noise of lasers and detectors

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 ...

Active control of the noise

International Nuclear Information System (INIS)

Rodriguez V, Luis Alfonso; Lopez Q, Jose German

2001-01-01

The problems of acoustic noise are more and more preponderant in the measure in that the amount of equipment and industrial machinery is increased such as fans, transformers, compressors etc. the use of devices passive mechanics for the reduction of the noise is effective and very appreciated because its effects embrace a wide range of acoustic frequency. However, to low frequencies, such devices become too big and expensive besides that present a tendency to do not effective. The control of active noise, CAN, using the electronic generation anti-noise, constitutes an interesting solution to the problem because their operation principle allows achieving an appreciable reduction of the noise by means of the use of compact devices. The traditional techniques for the control of acoustic noise like barriers and silenced to attenuate it, are classified as passive and their works has been accepted as norm as for the treatment of problems of noise it refers. Such techniques are considered in general very effective in the attenuation of noise of wide band. However, for low frequency, the required passive structures are too big and expensive; also, their effectiveness diminishes flagrantly, that which makes them impractical in many applications. The active suppression is profiled like a practical alternative for the reduction of acoustic noise. The idea in the active treatment of the noise it contemplates the use of a device electro-acoustic, like a speaker for example that it cancels to the noise by the generation of sounds of Same width and of contrary phase (anti-noise). The cancellation phenomenon is carried out when the ant-noise combines acoustically with the noise, what is in the cancellation of both sounds. The effectiveness of the cancellation of the primary source of noise depends on the precision with which the width and the phase of the generated ant-noise are controlled. The active control of noise, ANC (activates noise control), it is being investigated for

Noise in position measurement by centroid calculation

International Nuclear Information System (INIS)

Volkov, P.

1996-01-01

The position of a particle trajectory in a gaseous (or semiconductor) detector can be measured by calculating the centroid of the induced charge on the cathode plane. The charge amplifiers attached to each cathode strip introduce noise which is added to the signal. This noise broadens the position resolution line. Our article gives an analytical tool to estimate the resolution broadening due to the noise per strip and the number of strips involved in the centroid calculation. It is shown that the position resolution increases faster than the square root of the number of strips involved. We also consider the consequence of added interstrip capacitors, intended to diminish the differential nonlinearity. It is shown that the position error increases slower than linearly with the interstrip capacities, due to the cancellation of correlated noise. The estimation we give, can be applied to calculations of position broadening other than the centroid finding. (orig.)

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...

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

Science.gov (United States)

Stephens, David

2011-01-01

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

The effects of noise reduction technologies on the acceptance of background noise.

Science.gov (United States)

Lowery, Kristy Jones; Plyler, Patrick N

2013-09-01

Directional microphones (D-Mics) and digital noise reduction (DNR) algorithms are used in hearing aids to reduce the negative effects of background noise on performance. Directional microphones attenuate sounds arriving from anywhere other than the front of the listener while DNR attenuates sounds with physical characteristics of noise. Although both noise reduction technologies are currently available in hearing aids, it is unclear if the use of these technologies in isolation or together affects acceptance of noise and/or preference for the end user when used in various types of background noise. The purpose of the research was to determine the effects of D-Mic, DNR, or the combination of D-Mic and DNR on acceptance of noise and preference when listening in various types of background noise. An experimental study in which subjects were exposed to a repeated measures design was utilized. Thirty adult listeners with mild sloping to moderately severe sensorineural hearing loss participated (mean age 67 yr). Acceptable noise levels (ANLs) were obtained using no noise reduction technologies, D-Mic only, DNR only, and the combination of the two technologies (Combo) for three different background noises (single-talker speech, speech-shaped noise, and multitalker babble) for each listener. In addition, preference rankings of the noise reduction technologies were obtained within each background noise (1 = best, 3 = worst). ANL values were significantly better for each noise reduction technology than baseline; and benefit increased significantly from DNR to D-Mic to Combo. Listeners with higher (worse) baseline ANLs received more benefit from noise reduction technologies than listeners with lower (better) baseline ANLs. Neither ANL values nor ANL benefit values were significantly affected by background noise type; however, ANL benefit with D-Mic and Combo was similar when speech-like noise was present while ANL benefit was greatest for Combo when speech spectrum noise was

Noise Depression of Parasitic Capacitance for Frequency Detection of Micromechanical Bulk Disk Resonator

DEFF Research Database (Denmark)

Tang, Meng; Cagliani, Alberto; Escouflaire, Marie

2010-01-01

the frequency noise of the system. A capacitor cancellation circuit is used to subtract the parasitic capacitor. Measurements are conducted before and after the cancellation, and results show that after cancellation, the anti resonance is suppressed and the frequency noise is decreased, thus decreasing...

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.

Loading Effect on Tire Noise Radiation

OpenAIRE

Cao, Rui; Bolton, J Stuart

2016-01-01

Noise radiated by tires is a prominent noise pollution source and it is affected by many different parameters. Here, the effect of static load on tire noise radiation in a laboratory environment was investigated. The measurement was conducted by using the Tire Pavement Testing Apparatus (TPTA), on which a loaded tire can be run at speeds up to 50 km/hr; the tire noise was measured using a nearfield microphone method. The tire loading was varied from 500 to 900 pounds, and several different co...

Applications of digital processing for noise removal from plasma diagnostics

International Nuclear Information System (INIS)

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

1985-01-01

The use of digital signal techniques for removal of noise components present in plasma diagnostic signals is discussed, particularly with reference to diamagnetic loop signals. These signals contain noise due to power supply ripple in addition to plasma characteristics. The application of noise canceling techniques, such as adaptive noise canceling and model-based estimation, will be discussed. The use of computer codes such as SIG is described. 19 refs., 5 figs

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.

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

Stability and Linear Quadratic Differential Games of Discrete-Time Markovian Jump Linear Systems with State-Dependent Noise

Directory of Open Access Journals (Sweden)

Huiying Sun

2014-01-01

Full Text Available We mainly consider the stability of discrete-time Markovian jump linear systems with state-dependent noise as well as its linear quadratic (LQ differential games. A necessary and sufficient condition involved with the connection between stochastic Tn-stability of Markovian jump linear systems with state-dependent noise and Lyapunov equation is proposed. And using the theory of stochastic Tn-stability, we give the optimal strategies and the optimal cost values for infinite horizon LQ stochastic differential games. It is demonstrated that the solutions of infinite horizon LQ stochastic differential games are concerned with four coupled generalized algebraic Riccati equations (GAREs. Finally, an iterative algorithm is presented to solve the four coupled GAREs and a simulation example is given to illustrate the effectiveness of it.

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.

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.

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.

Diagrammatic cancellations and the gauge dependence of QED

Energy Technology Data Exchange (ETDEWEB)

Kißler, Henry, E-mail: kissler@physik.hu-berlin.de [Department of Mathematical Sciences, University of Liverpool, L69 7ZL, Liverpool (United Kingdom); Department of Mathematics, Humboldt-Universität zu Berlin, Rudower Chaussee 25, D-12489 Berlin (Germany); Kreimer, Dirk, E-mail: kreimer@math.hu-berlin.de [Department of Mathematics, Humboldt-Universität zu Berlin, Rudower Chaussee 25, D-12489 Berlin (Germany)

2017-01-10

This letter examines diagrammatic cancellations for Quantum Electrodynamics (QED) in the general linear gauge. These cancellations combine Feynman graphs of various topologies and provide a method to reconstruct the gauge dependence of the electron propagator from the result of a particular gauge by means of a linear Dyson–Schwinger equation. We use this method in combination with dimensional regularization to demonstrate how the 3-loop ε-expansion in the Feynman gauge determines the ε-expansions for all gauge parameter dependent terms to 4 loops.

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.

Noise from Two-Blade Propellers

Science.gov (United States)

Stowell, E Z; Deming, A F

1936-01-01

The two-blade propeller, one of the most powerful sources of sound known, has been studied with the view of obtaining fundamental information concerning the noise emission. In order to eliminate engine noise, the propeller was mounted on an electric motor. A microphone was used to pick up the sound whose characteristics were studied electrically. The distribution of noise throughout the frequency range, as well as the spatial distribution about the propeller, was studied. The results are given in the form of polar diagrams. An appendix of common acoustical terms is included.

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....

Noise Reduction with Optimal Variable Span Linear Filters

DEFF Research Database (Denmark)

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

2016-01-01

In this paper, the problem of noise reduction is addressed as a linear filtering problem in a novel way by using concepts from subspace-based enhancement methods, resulting in variable span linear filters. This is done by forming the filter coefficients as linear combinations of a number...... included in forming the filter. Using these concepts, a number of different filter designs are considered, like minimum distortion, Wiener, maximum SNR, and tradeoff filters. Interestingly, all these can be expressed as special cases of variable span filters. We also derive expressions for the speech...... demonstrate the advantages and properties of the variable span filter designs, and their potential performance gain compared to widely used speech enhancement methods....

Hearing aid noise suppression and working memory function

OpenAIRE

Fischer, Rosa-Linde; Neher, Tobias; Wagener, Kirsten C.

2017-01-01

OBJECTIVE: Research findings concerning the relation between benefit from hearing aid (HA) noise suppression and working memory function are inconsistent. The current study thus investigated the effects of three noise suppression algorithms on auditory working memory and the relation with reading span.DESIGN: Using a computer simulation of bilaterally fitted HAs, four settings were tested: (1) unprocessed, (2) directional microphones, (3) single-channel noise reduction and (4) binaural cohere...

A Novel Offset Cancellation Based on Parasitic-Insensitive Switched-Capacitor Sensing Circuit for the Out-of-Plane Single-Gimbaled Decoupled CMOS-MEMS Gyroscope

Science.gov (United States)

Chang, Ming-Hui; Huang, Han-Pang

2013-01-01

This paper presents a novel parasitic-insensitive switched-capacitor (PISC) sensing circuit design in order to obtain high sensitivity and ultra linearity and reduce the parasitic effect for the out-of-plane single-gimbaled decoupled CMOS-MEMS gyroscope (SGDG). According to the simulation results, the proposed PISC circuit has better sensitivity and high linearity in a wide dynamic range. Experimental results also show a better performance. In addition, the PISC circuit can use signal processing to cancel the offset and noise. Thus, this circuit is very suitable for gyroscope measurement. PMID:23493122

Noise analysis and performance of a selfscanned linear InSb detector array

International Nuclear Information System (INIS)

Finger, G.; Meyer, M.; Moorwood, A.F.M.

1987-01-01

A noise model for detectors operated in the capacitive discharge mode is presented. It is used to analyze the noise performance of the ESO nested timing readout technique applied to a linear 32-element InSb array which is multiplexed by a silicon switched-FET shift register. Analysis shows that KTC noise of the videoline is the major noise contribution; it can be eliminated by weighted double-correlated sampling. Best noise performance of this array is achieved at the smallest possible reverse bias voltage (not more than 20 mV) whereas excess noise is observed at higher reverse bias voltages. 5 references

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...

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.

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...

Multiple-energy tissue-cancellation applications of a digital beam attenuator to chest radiography

International Nuclear Information System (INIS)

Dobbins, J.T. III.

1985-01-01

The digitally-formed primary beam attenuator (DBA) spatially modulates the x-ray fluence incident upon the patient to selectively attenuate regions of interest. The DBA attenuating mask is constructed from CeO 2 powder by a modified printing technique and uses image information from an initial low-dose exposure. Two tissue-cancellation imaging techniques are investigated with the DBA: (1) energy-dependent information is used to form a beam attenuator that attenuates specific tissues in the primary x-ray beam for tissue-cancelled film radiography; (2) the beam attenuator is used to improve image signal-to-noise and scattered radiation properties in traditional energy-subtraction tissue-cancellation imaging with digital detectors. The tissue-cancellation techniques in the primary x-ray beam were capable of adequately removing either soft-tissue or bone from the final compensated film radiograph when using a phantom with well defined soft-tissue and bone sections. However, when tried on an anthropomorphic chest phantom the results were adequate for cancellation of large soft tissue structures, but unsatisfactory for cancellation of bony structures such as the ribs, because of the limited spatial frequency content of the attenuating mask. The second technique (with digital detectors) showed improved uniformity of image signal-to-noise and a two-fold increase in soft-tissue nodule contrast due to improved scattered radiation properties. The tissue-cancelled images contained residual image contributions from the presence of the attenuating mask, but this residual may be correctable by future algorithms

Noise analysis of fluid-valve system in a linear compressor using CAE

International Nuclear Information System (INIS)

Lee, Jun Ho; Jeong, Weui Bong; Kim, Dang Ju

2009-01-01

A linear compressor in a refrigerator uses piston motion to transfer refrigerant so its efficiency is higher than a previous reciprocal compressor. Because of interaction between refrigerant and valves system in the linear compressor, however, noise has been a main issue. In spite of doing many experimental researches, there is no way to rightly predict the noise. In order to solve this limitation, the CAE analysis is applied. For giving credit to these computational data, all of the data are experimentally validated.

Analysis of noise emitted from diesel engines

Science.gov (United States)

Narayan, S.

2015-12-01

In this work combustion noise produced in diesel engines has been investigated. In order to reduce the exhaust emissions various injection parameters need to be studied and optimized. The noise has been investigated by mean of data obtained from cylinder pressure measurements using piezo electric transducers and microphones on a dual cylinder diesel engine test rig. The engine was run under various operating conditions varying various injection parameters to investigate the effects of noise emissions under various testing conditions.

DART Core/Combustor-Noise Initial Test Results

Science.gov (United States)

Boyle, Devin K.; Henderson, Brenda S.; Hultgren, Lennart S.

2017-01-01

Contributions from the combustor to the overall propulsion noise of civilian transport aircraft are starting to become important due to turbofan design trends and advances in mitigation of other noise sources. Future propulsion systems for ultra-efficient commercial air vehicles are projected to be of increasingly higher bypass ratio from larger fans combined with much smaller cores, with ultra-clean burning fuel-flexible combustors. Unless effective noise-reduction strategies are developed, combustor noise is likely to become a prominent contributor to overall airport community noise in the future. The new NASA DGEN Aero0propulsion Research Turbofan (DART) is a cost-efficient testbed for the study of core-noise physics and mitigation. This presentation gives a brief description of the recently completed DART core combustor-noise baseline test in the NASA GRC Aero-Acoustic Propulsion Laboratory (AAPL). Acoustic data was simultaneously acquired using the AAPL overhead microphone array in the engine aft quadrant far field, a single midfield microphone, and two semi-infinite-tube unsteady pressure sensors at the core-nozzle exit. An initial assessment shows that the data is of high quality and compares well with results from a quick 2014 feasibility test. Combustor noise components of measured total-noise signatures were educed using a two-signal source-separation method an dare found to occur in the expected frequency range. The research described herein is aligned with the NASA Ultra-Efficient Commercial Transport strategic thrust and is supported by the NASA Advanced Air Vehicle Program, Advanced Air Transport Technology Project, under the Aircraft Noise Reduction Subproject.

Impact of noise and other factors on speech recognition in anaesthesia

DEFF Research Database (Denmark)

Alapetite, Alexandre

2008-01-01

of training. Methods: Eight volunteers read aloud a total of about 3 600 typical short anaesthesia comments to be transcribed by a continuous speech recognition system. Background noises were collected in an operating room and reproduced. A regression analysis and descriptive statistics were done to evaluate...... operations. Objective: The aim of the experiment is to evaluate the relative impact of several factors affecting speech recognition when used in operating rooms, such as the type or loudness of background noises, type of microphone, type of recognition mode (free speech versus command mode), and type...... the relative effect of various factors. Results: Some factors have a major impact, such as the words to be recognised, the type of recognition, and participants. The type of microphone is especially significant when combined with the type of noise. While loud noises in the operating room can have a predominant...

Influence of Gaussian white noise on the frequency-dependent linear polarizability of doped quantum dot

International Nuclear Information System (INIS)

Ganguly, Jayanta; Ghosh, Manas

2014-01-01

Highlights: • Linear polarizability of quantum dot has been studied. • Quantum dot is doped with a repulsive impurity. • The polarizabilities are frequency-dependent. • Influence of Gaussian white noise has been monitored. • Noise exploited is of additive and multiplicative nature. - Abstract: We investigate the profiles of diagonal components of frequency-dependent linear (α xx and α yy ) optical response of repulsive impurity doped quantum dots. The dopant impurity potential chosen assumes Gaussian form. The study principally puts emphasis on investigating the role of noise on the polarizability components. In view of this we have exploited Gaussian white noise containing additive and multiplicative characteristics (in Stratonovich sense). The frequency-dependent polarizabilities are studied by exposing the doped dot to a periodically oscillating external electric field of given intensity. The oscillation frequency, confinement potentials, dopant location, and above all, the noise characteristics tune the linear polarizability components in a subtle manner. Whereas the additive noise fails to have any impact on the polarizabilities, the multiplicative noise influences them delicately and gives rise to additional interesting features

A novel full scale experimental characterization of wind turbine aero-acoustic noise sources - preliminary results

DEFF Research Database (Denmark)

Aagaard Madsen, Helge; Bertagnolio, Franck; Fischer, Andreas

2016-01-01

of the blade and the noise on the ground in a distance of about one rotor diameter. In total six surface microphones were used to measure the SP at the leading edge (LE) and trailing edge (TE) of the blade. In parallel noise was measured by eight microphones placed on plates on the ground around the turbine......The paper describes a novel full scale experiment on a 500 kW wind turbine with the main objective to characterize the aero-acoustic noise sources. The idea behind the instrumentation is to study the link and correlation between the surface pressure (SP) fluctuations in the boundary layer...... in equidistant angles on a circle with a radius of about one rotor diameter. The data were analyzed in segments of 2.2 s which is the time for one rotor revolution. The spectra for the TE microphones on the suction side of the blade show a characteristic roll-off pattern around a frequency of 600-700 Hz...

Optimal Linear Filters for Pulse Height Measurements in the Presence of Noise

International Nuclear Information System (INIS)

Nygaard, K.

1966-07-01

For measurements of nuclear pulse height spectra a linear filter is used between the pulse amplifier and the pulse height recorder so as to improve the signal/noise ratio. The problem of finding the optimal filter is investigated with emphasis on technical realizability. The maximum available signal/noise ratio is theoretically calculated on the basis of all the information which can be found in the output of the pulse amplifier, and on an assumed a priori knowledge of the pulse time of arrival. It is then shown that the maximum available signal/noise ratio can be obtained with practical measurements without any a priori knowledge of pulse time of arrival, and a general description of the optimal linear filter is given. The solution is unique, technically realizable, and based solely on data (noise power spectrum and pulse shape) which can be measured at the output terminals of the pulse amplifier used

Optimal Linear Filters for Pulse Height Measurements in the Presence of Noise

Energy Technology Data Exchange (ETDEWEB)

Nygaard, K

1966-07-15

For measurements of nuclear pulse height spectra a linear filter is used between the pulse amplifier and the pulse height recorder so as to improve the signal/noise ratio. The problem of finding the optimal filter is investigated with emphasis on technical realizability. The maximum available signal/noise ratio is theoretically calculated on the basis of all the information which can be found in the output of the pulse amplifier, and on an assumed a priori knowledge of the pulse time of arrival. It is then shown that the maximum available signal/noise ratio can be obtained with practical measurements without any a priori knowledge of pulse time of arrival, and a general description of the optimal linear filter is given. The solution is unique, technically realizable, and based solely on data (noise power spectrum and pulse shape) which can be measured at the output terminals of the pulse amplifier used.

Design and implementation of efficient low complexity biomedical artifact canceller for nano devices

Directory of Open Access Journals (Sweden)

Md Zia Ur RAHMAN

2016-07-01

Full Text Available In the current day scenario, with the rapid development of communication technology remote health care monitoring becomes as an intense research area. In remote health care monitoring, the primary aim is to facilitate the doctor with high resolution biomedical data. In order to cancel various artifacts in clinical environment in this paper we propose some efficient adaptive noise cancellation techniques. To obtain low computational complexity we combine clipping the data or error with Least Mean Square (LMS algorithm. This results sign regressor LMS (SRLMS, sign LMS (SLMS and sign LMS (SSLMS algorithms. Using these algorithms, we design Very-large-scale integration (VLSI architectures of various Biomedical Noise Cancellers (BNCs. In addition, the filtering capabilities of the proposed implementations are measured using real biomedical signals. Among the various BNCs tested, SRLMS based BNC is found to be better with reference to convergence speed, filtering capability and computational complexity. The main advantage of this technique is it needs only one multiplication to compute next weight. In this manner SRLMS based BNC is independent of filter length with reference to its computations. Whereas, the average signal to noise ratio achieved in the noise cancellation experiments are recorded as 7.1059dBs, 7.1776dBs, 6.2795dBs and 5.8847dBs for various BNCs based on LMS, SRLMS, SLMS and SSSLMS algorithms respectively. Based on the filtering characteristics, convergence and computational complexity, the proposed SRLMS based BNC architecture is well suited for nanotechnology applications.

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...

Comparison of parametric and linear mass detection in the presence of detection noise

International Nuclear Information System (INIS)

Yie, Zi; Zielke, Mark A; Burgner, Christopher B; Turner, Kimberly L

2011-01-01

We experimentally investigate the performance of a nonlinear parametrically driven mass sensor in the presence of detection noise. Mass detection is achieved by measuring the amount of methanol vapor adsorption on the sensor. To demonstrate the advantage of parametric sensing in counteracting the influence of detection noise, we operate the sensor in both the parametric and harmonic resonance mode. Comparison of the results shows that in contrast to conventional linear harmonic sensing, the detection sensitivity does not deteriorate for the parametric case when a tenfold increase in detection noise is introduced. Furthermore, we demonstrate additional functionality of the parametric sensor by utilizing it as a threshold detector, whose performance remains the same despite the added detection noise. Taken together, these results suggest that for mass detection in the presence of detection noise, a parametrically operated sensor may offer better performance over one operated harmonically in the linear regime.

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

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.)

Stability of the trivial solution for linear stochastic differential equations with Poisson white noise

International Nuclear Information System (INIS)

Grigoriu, Mircea; Samorodnitsky, Gennady

2004-01-01

Two methods are considered for assessing the asymptotic stability of the trivial solution of linear stochastic differential equations driven by Poisson white noise, interpreted as the formal derivative of a compound Poisson process. The first method attempts to extend a result for diffusion processes satisfying linear stochastic differential equations to the case of linear equations with Poisson white noise. The developments for the method are based on Ito's formula for semimartingales and Lyapunov exponents. The second method is based on a geometric ergodic theorem for Markov chains providing a criterion for the asymptotic stability of the solution of linear stochastic differential equations with Poisson white noise. Two examples are presented to illustrate the use and evaluate the potential of the two methods. The examples demonstrate limitations of the first method and the generality of the second method

Estimating integrated variance in the presence of microstructure noise using linear regression

Science.gov (United States)

Holý, Vladimír

2017-07-01

Using financial high-frequency data for estimation of integrated variance of asset prices is beneficial but with increasing number of observations so-called microstructure noise occurs. This noise can significantly bias the realized variance estimator. We propose a method for estimation of the integrated variance robust to microstructure noise as well as for testing the presence of the noise. Our method utilizes linear regression in which realized variances estimated from different data subsamples act as dependent variable while the number of observations act as explanatory variable. We compare proposed estimator with other methods on simulated data for several microstructure noise structures.

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...

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...

Noise Source Visualization Using a Digital Voice Recorder and Low-Cost Sensors.

Science.gov (United States)

Cho, Yong Thung

2018-04-03

Accurate sound visualization of noise sources is required for optimal noise control. Typically, noise measurement systems require microphones, an analog-digital converter, cables, a data acquisition system, etc., which may not be affordable for potential users. Also, many such systems are not highly portable and may not be convenient for travel. Handheld personal electronic devices such as smartphones and digital voice recorders with relatively lower costs and higher performance have become widely available recently. Even though such devices are highly portable, directly implementing them for noise measurement may lead to erroneous results since such equipment was originally designed for voice recording. In this study, external microphones were connected to a digital voice recorder to conduct measurements and the input received was processed for noise visualization. In this way, a low cost, compact sound visualization system was designed and introduced to visualize two actual noise sources for verification with different characteristics: an enclosed loud speaker and a small air compressor. Reasonable accuracy of noise visualization for these two sources was shown over a relatively wide frequency range. This very affordable and compact sound visualization system can be used for many actual noise visualization applications in addition to educational purposes.

An inverter-based capacitive trans-impedance amplifier readout with offset cancellation and temporal noise reduction for IR focal plane array

Science.gov (United States)

Chen, Hsin-Han; Hsieh, Chih-Cheng

2013-09-01

This paper presents a readout integrated circuit (ROIC) with inverter-based capacitive trans-impedance amplifier (CTIA) and pseudo-multiple sampling technique for infrared focal plane array (IRFPA). The proposed inverter-based CTIA with a coupling capacitor [1], executing auto-zeroing technique to cancel out the varied offset voltage from process variation, is used to substitute differential amplifier in conventional CTIA. The tunable detector bias is applied from a global external bias before exposure. This scheme not only retains stable detector bias voltage and signal injection efficiency, but also reduces the pixel area as well. Pseudo-multiple sampling technique [2] is adopted to reduce the temporal noise of readout circuit. The noise reduction performance is comparable to the conventional multiple sampling operation without need of longer readout time proportional to the number of samples. A CMOS image sensor chip with 55×65 pixel array has been fabricated in 0.18um CMOS technology. It achieves a 12um×12um pixel size, a frame rate of 72 fps, a power-per-pixel of 0.66uW/pixel, and a readout temporal noise of 1.06mVrms (16 times of pseudo-multiple sampling), respectively.

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)

Effects of noise, nonlinear processing, and linear filtering on perceived music quality.

Science.gov (United States)

Arehart, Kathryn H; Kates, James M; Anderson, Melinda C

2011-03-01

The purpose of this study was to determine the relative impact of different forms of hearing aid signal processing on quality ratings of music. Music quality was assessed using a rating scale for three types of music: orchestral classical music, jazz instrumental, and a female vocalist. The music stimuli were subjected to a wide range of simulated hearing aid processing conditions including, (1) noise and nonlinear processing, (2) linear filtering, and (3) combinations of noise, nonlinear, and linear filtering. Quality ratings were measured in a group of 19 listeners with normal hearing and a group of 15 listeners with sensorineural hearing impairment. Quality ratings in both groups were generally comparable, were reliable across test sessions, were impacted more by noise and nonlinear signal processing than by linear filtering, and were significantly affected by the genre of music. The average quality ratings for music were reasonably well predicted by the hearing aid speech quality index (HASQI), but additional work is needed to optimize the index to the wide range of music genres and processing conditions included in this study.

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

Elimination of noise peak for signal processing in Johnson noise thermometry development

International Nuclear Information System (INIS)

Hwang, I. G.; Moon, B. S.; Jeong, J. E.; Jeo, Y. H.; Kisner, Roger A.

2003-01-01

The internal and external noise is the most considering obstacle in development of Johnson Noise Thermometry system. This paper addresses an external noise elimination issue of the Johnson Noise Thermometry system which is underway of development in collaboration between KAERI and ORNL. Although internal random noise is canceled by Cross Power Spectral Density function, a continuous wave penetrating into the electronic circuit is eliminated by the difference of peaks between Johnson signal and external noise. The elimination logic using standard deviation of CPSD and energy leakage problem in discrete CPSD function are discussed in this paper

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

High phase noise tolerant pilot-tone-aided DP-QPSK optical communication systems

DEFF Research Database (Denmark)

Zhang, Xu; Pang, Xiaodan; Deng, Lei

2012-01-01

In this paper we experimentally demonstrate a novel, high phase-noise tolerant, optical dual polarization (DP) quadrature phase-shift keying (QPSK) communication system based on pilot-tone-aided phase noise cancellation (PNC) algorithm. Vertical cavity surface emitting lasers (VCSELs) with approx......In this paper we experimentally demonstrate a novel, high phase-noise tolerant, optical dual polarization (DP) quadrature phase-shift keying (QPSK) communication system based on pilot-tone-aided phase noise cancellation (PNC) algorithm. Vertical cavity surface emitting lasers (VCSELs...

On estimation of the noise variance in high-dimensional linear models

OpenAIRE

Golubev, Yuri; Krymova, Ekaterina

2017-01-01

We consider the problem of recovering the unknown noise variance in the linear regression model. To estimate the nuisance (a vector of regression coefficients) we use a family of spectral regularisers of the maximum likelihood estimator. The noise estimation is based on the adaptive normalisation of the squared error. We derive the upper bound for the concentration of the proposed method around the ideal estimator (the case of zero nuisance).

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.

A two-microphone noise reduction system for cochlear implant users with nearby microphones. Part II: Performance Evaluation

OpenAIRE

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

2008-01-01

Users of cochlear implants (auditory aids, which stimulate the auditory nerve electrically at the inner ear) often suffer from poor speech understanding in noise. We evaluate a small (intermicrophone distance 7 mm) and computationally inexpensive adaptive noise reduction system suitable for behind-the-ear cochlear implant speech processors. The system is evaluated in simulated and real, anechoic and reverberant environments. Results from simulations show improvements of 3.4 to 9.3 dB in signa...

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).

Beamforming design with proactive interference cancelation in MISO interference channels

Science.gov (United States)

Li, Yang; Tian, Yafei; Yang, Chenyang

2015-12-01

In this paper, we design coordinated beamforming at base stations (BSs) to facilitate interference cancelation at users in interference networks, where each BS is equipped with multiple antennas and each user is with a single antenna. By assuming that each user can select the best decoding strategy to mitigate the interference, either canceling the interference after decoding when it is strong or treating it as noise when it is weak, we optimize the beamforming vectors that maximize the sum rate for the networks under different interference scenarios and find the solutions of beamforming with closed-form expressions. The inherent design principles are then analyzed, and the performance gain over passive interference cancelation is demonstrated through simulations in heterogeneous cellular networks.

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

Drone noise

Science.gov (United States)

Tinney, Charles; Sirohi, Jayant; University of Texas at Austin Team

2017-11-01

A basic understanding of the noise produced by single and multirotor drones operating at static thrust conditions is presented. This work acts as an extension to previous efforts conducted at The University of Texas at Austin (Tinney et al. 2017, AHS Forum 73). Propeller diameters ranging from 8 inch to 12 inch are examined for configurations comprising an isolated rotor, a quadcopter configuration and a hexacopter configuration, and with a constant drone pitch of 2.25. An azimuthal array of half-inch microphones, placed between 2 and 3 hub-center diameters from the drone center, are used to assess the acoustic near-field. Thrust levels, acquired using a six degree-of-freedom load cell, are then used to correlate acoustic noise levels to aerodynamic performance for each drone configuration. The findings reveal a nearly logarithmic increase in noise with increasing thrust. However, for the same thrust condition, considerable noise reduction is achieved by increasing the number of propeller blades thereby reducing the blade passage frequency and both the thickness and loading noise sources that accompany it.

Reduction of environmental MHz noise for SQUID application

Energy Technology Data Exchange (ETDEWEB)

Araya, T. [Graduate School of Engineering Science, Osaka University, 1-3 Machikaneyama, Toyonaka, Osaka 560-8531 (Japan)]. E-mail: araya@sup.ee.es.osaka-u.ac.jp; Kitamura, Y. [Graduate School of Engineering Science, Osaka University, 1-3 Machikaneyama, Toyonaka, Osaka 560-8531 (Japan); Kamishiro, M. [Graduate School of Engineering Science, Osaka University, 1-3 Machikaneyama, Toyonaka, Osaka 560-8531 (Japan); Sakuta, K. [Graduate School of Engineering Science, Osaka University, 1-3 Machikaneyama, Toyonaka, Osaka 560-8531 (Japan); Itozaki, H. [Graduate School of Engineering Science, Osaka University, 1-3 Machikaneyama, Toyonaka, Osaka 560-8531 (Japan)]. E-mail: itozaki@ee.es.osaka-u.ac.jp

2006-10-01

It is important to remove large environmental noise in measurement using SQUIDs without magnetic shielding. Active noise control (ANC) is an effective method to remove the environmental noise. The environmental noise has been reduced by the ANC system in the radio frequency region around MHz. The anti-phase waves of the environmental noise should be generated by this system. The ANC system including the phase and amplitude control circuit was developed to make the anti-phase waves in the MHz region. In this paper, sinusoidal waves with a MHz frequency were used as the environmental noise. When a coil antenna was used for a receiver antenna, this ANC system suppressed these sinusoidal waves to the white noise level about 40 dB. When we used a SQUID as a receiver antenna, we also cancelled sinusoidal waves to the white noise level by this system. This shows that the ANC system is useful to reduce an environmental noise when this ANC system is developed to cancel multi-frequency noise.

Reduction of environmental MHz noise for SQUID application

International Nuclear Information System (INIS)

Araya, T.; Kitamura, Y.; Kamishiro, M.; Sakuta, K.; Itozaki, H.

2006-01-01

It is important to remove large environmental noise in measurement using SQUIDs without magnetic shielding. Active noise control (ANC) is an effective method to remove the environmental noise. The environmental noise has been reduced by the ANC system in the radio frequency region around MHz. The anti-phase waves of the environmental noise should be generated by this system. The ANC system including the phase and amplitude control circuit was developed to make the anti-phase waves in the MHz region. In this paper, sinusoidal waves with a MHz frequency were used as the environmental noise. When a coil antenna was used for a receiver antenna, this ANC system suppressed these sinusoidal waves to the white noise level about 40 dB. When we used a SQUID as a receiver antenna, we also cancelled sinusoidal waves to the white noise level by this system. This shows that the ANC system is useful to reduce an environmental noise when this ANC system is developed to cancel multi-frequency noise

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...

Communication system with adaptive noise suppression

Science.gov (United States)

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

2007-01-01

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

Noise Source Visualization Using a Digital Voice Recorder and Low-Cost Sensors

Directory of Open Access Journals (Sweden)

Yong Thung Cho

2018-04-01

Full Text Available Accurate sound visualization of noise sources is required for optimal noise control. Typically, noise measurement systems require microphones, an analog-digital converter, cables, a data acquisition system, etc., which may not be affordable for potential users. Also, many such systems are not highly portable and may not be convenient for travel. Handheld personal electronic devices such as smartphones and digital voice recorders with relatively lower costs and higher performance have become widely available recently. Even though such devices are highly portable, directly implementing them for noise measurement may lead to erroneous results since such equipment was originally designed for voice recording. In this study, external microphones were connected to a digital voice recorder to conduct measurements and the input received was processed for noise visualization. In this way, a low cost, compact sound visualization system was designed and introduced to visualize two actual noise sources for verification with different characteristics: an enclosed loud speaker and a small air compressor. Reasonable accuracy of noise visualization for these two sources was shown over a relatively wide frequency range. This very affordable and compact sound visualization system can be used for many actual noise visualization applications in addition to educational purposes.

An acoustic vector based approach to locate low frequency noise sources in 3D

NARCIS (Netherlands)

Bree, H.-E. de; Ostendorf, C.; Basten, T.

2009-01-01

Although low frequency noise is an issue of huge societal importance, traditional acoustic testing methods have limitations in finding the low frequency source. It is hard to determine the direction of the noise using traditional microphones. Three dimensional sound probes capturing the particle

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.

Imperfect generalized transmit beamforming with co-channel interference cancelation

KAUST Repository

Radaydeh, Redha Mahmoud Mesleh

2010-10-01

The performance of a generalized single-stream transmit beamforming scheme employing receive co-channel interference -steering algorithms in slowly varying and flat fading channels is analyzed. The impact of imperfect prediction of channel state information (CSI) for the desired user spatially uncorrelated transmit channels is considered. Both dominant interference cancelation and adaptive arbitrary interference cancelation algorithms for closely spaced receive antennas are used. The impact of outdated statistical ordering of the interferers instantaneous powers on the effectiveness of dominant interference cancelation is investigated against the less complex adaptive arbitrary cancelation scheme. For the system models described above, new exact formulas for the statistics of combined signal-to-interference-plus-noise ratio (SINR) are derived, from which results for conventional maximum ratio transmission (MRT) and best transmit channel selection schemes can be deduced as limiting cases. The results presented herein can be used to obtain quantitative measure for various performance metrics, and in addition to investigate the performance-complexity tradeoff for different multiple-antenna system models. © 2010 IEEE.

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....

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...

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...

A particle velocity sensor to measure the sound from a structure in the presence of background noise

NARCIS (Netherlands)

de Bree, H.E.; Druyvesteyn, W.F.

2005-01-01

The performance (or quality) of a product is often checked by measuring the radiated sound (noise) from the vibrating structure. Often this test has to be done in an environment with background noise, which makes the measurement difficult. When using a (pressure) microphone the background noise can

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.

Linearizing of Low Noise Power Amplifier Using 5.8GHz Double Loop Feedforward Linearization Technique

Directory of Open Access Journals (Sweden)

Abdulkareem Mokif Obais

2017-05-01

Full Text Available In this paper, a double loop feedforward linearization technique is analyzed and built with a MMIC low noise amplifier “HMC753” as main amplifier and a two-stage class-A power amplifier as error amplifier. The system is operated with 5V DC supply at a center frequency of 5.8GHz and a bandwidth of 500MHz. The proposed technique, increases the linearity of the MMIC amplifier from 18dBm at 1dB compression point to more than 26dBm. In addition, the proposed system is tested with OFDM signal and it reveals good response in maximizing the linearity region and eliminating distortions. The proposed system is designed and simulated onAdvanced Wave Research-Microwave Office (AWR-MWO.

A conceptual framework for noise reduction

CERN Document Server

Benesty, Jacob

2015-01-01

Though noise reduction and speech enhancement problems have been studied for at least five decades, advances in our understanding and the development of reliable algorithms are more important than ever, as they support the design of tailored solutions for clearly defined applications. In this work, the authors propose a conceptual framework that can be applied to the many different aspects of noise reduction, offering a uniform approach to monaural and binaural noise reduction problems, in the time domain and in the frequency domain, and involving a single or multiple microphones. Moreover, the derivation of optimal filters is simplified, as are the performance measures used for their evaluation.

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...

Active Control of Fan Noise-Feasibility Study. Volume 2: Canceling Noise Source-Design of an Acoustic Plate Radiator Using Piezoceramic Actuators

Science.gov (United States)

Pla, F. G.; Rajiyah, H.

1995-01-01

The feasibility of using acoustic plate radiators powered by piezoceramic thin sheets as canceling sources for active control of aircraft engine fan noise is demonstrated. Analytical and numerical models of actuated beams and plates are developed and validated. An optimization study is performed to identify the optimum combination of design parameters that maximizes the plate volume velocity for a given resonance frequency. Fifteen plates with various plate and actuator sizes, thicknesses, and bonding layers were fabricated and tested using results from the optimization study. A maximum equivalent piston displacement of 0.39 mm was achieved with the optimized plate samples tested with only one actuator powered, corresponding to a plate deflection at the center of over 1 millimeter. This is very close to the deflection required for a full size engine application and represents a 160-fold improvement over previous work. Experimental results further show that performance is limited by the critical stress of the piezoceramic actuator and bonding layer rather than by the maximum moment available from the actuator. Design enhancements are described in detail that will lead to a flight-worthy acoustic plate radiator by minimizing actuator tensile stresses and reducing nonlinear effects. Finally, several adaptive tuning methods designed to increase the bandwidth of acoustic plate radiators are analyzed including passive, active, and semi-active approaches. The back chamber pressurization and volume variation methods are investigated experimentally and shown to be simple and effective ways to obtain substantial control over the resonance frequency of a plate radiator. This study shows that piezoceramic-based plate radiators can be a viable acoustic source for active control of aircraft engine fan noise.

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.

Noise from wind turbines

International Nuclear Information System (INIS)

Andersen, B.; Larsen, P.

1993-01-01

Denmark has 3200 wind turbines with an installed maximum capacity of 418MW. The most important Danish research projects into wind turbine noise and the main results are listed. These date from 1983. Two comprehensive studies are currently in progress. The first is an analytical and empirical investigation of aerodynamic noise from wind turbine rotors and has so far dealt mainly with tip noise. The measurement method, using a hard board mounted microphone on the ground near the turbine, is described. Four different tip designs have been tested. Some examples of reference sound power level spectra for three of the designs are presented. During the past two years a computerbased data acquisition system has been used for real-time determination of sound power levels. The second study, which has just commenced, is on annoyance from wind turbine noise. It will include noise measurements, masking calculations and a social survey on the perceived nuisance. (UK)

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...

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...

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....

Relation between nonlinear or 'not-linear' characteristics in nuclear kinetics and noise analysis of neutron flux

International Nuclear Information System (INIS)

Kataoka, H.

1975-01-01

The 'not-linear' or '2nd-class-nonlinear' characteristics in nuclear reactor kinetics with the feedback effect in the high-power operation and induce the increase in the amplitude of the neutron flux noise, specially in the very low frequency region. The fundamental behaviour of 'not-linear' characteristics and its effect for the reactor noise was investigated. Application of the reactor noise analysis technique to power reactors has not been successful because of unknown large disagreement between the result of the conventional theoretical analysis and the experimental facts. When the cause of this discrepancy is clear, reactor noise analysis techniques can be effectively applied to instrumentation, control, monitoring and diagnosis of power reactors. (author)

40 CFR 201.23 - Test site, weather conditions and background noise criteria for measurement at a 30 meter (100...

Science.gov (United States)

2010-07-01

... background noise criteria for measurement at a 30 meter (100 feet) distance of the noise from locomotive and... and background noise criteria for measurement at a 30 meter (100 feet) distance of the noise from... meters (4 feet) above the ground at the microphone location, except as provided in paragraph (c) of this...

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...

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...

Noise immission from wind turbines

International Nuclear Information System (INIS)

1999-01-01

The project has dealt with practical ways to reduce the influence of background noise caused by wind acting on the measuring microphones. The uncertainty of measured noise emission (source strength) has been investigated. The main activity was a Round Robin Test involving measurements by five laboratories at the same wind turbine. Each laboratory brought its own instrumentation and performed the measurements and analyses according to their interpretation. The tonality of wind turbine noise is an essential component of the noise impact on the environment. In the present project the uncertainty in the newest existing methods for assessing tonality was investigated. The project included noise propagation measurements in different weather conditions around wind turbines situated in different types of terrain. The results were used to validate a noise propagation model developed in the project. Finally, the project also included a study with listeners evaluating recordings of wind turbine noise. The results are intended as guidance for wind turbine manufacturers in identifying the aspects of wind turbine noise most important to annoyance. (author)

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

STATE OF THE ART TECHNIQUES USED FOR NOISE SOURCE IDENTIFICATION ON COMPLEX BODIES

Directory of Open Access Journals (Sweden)

Corneliu STOICA

2010-03-01

Full Text Available Over the last few decades, many approaches have been undertaken in order to asses detailed noise source identification on complex bodies, i.e. aircrafts, cars, machinery. Noise source identification implies to accurately obtain the position and frequency of the dominant noise sources. There are cases where traditional testing methods can not be applied at all or their use involves some limitations. Optical systems used for near field analysis require a line of sight that may not be available. The state-of-the-art technology for this purpose is the use of a large number of microphones whose signals are acquired simultaneously, i.e. microphone phased array. Due to the excessive cost of the instruments and the data acquisition system required, the implementation of this technology was restricted to governmental agencies (NASA, DLR and big companies such as Boeing and Airbus. During the past years, this technique was developed in wind tunnels and some universities to perform noise source identification on scale airframes, main landing gear models, and aerodynamic profiles (used on airplanes, helicopter rotors and wind mills.

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

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

Radiation impedance of condenser microphones and their diffuse-field responses

DEFF Research Database (Denmark)

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

2010-01-01

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

Static pressure and temperature coefficients of laboratory standard microphones

DEFF Research Database (Denmark)

Rasmussen, Knud

1996-01-01

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

Fabricating off-diagonal components of frequency-dependent linear and nonlinear polarizabilities of doped quantum dots by Gaussian white noise

International Nuclear Information System (INIS)

Saha, Surajit; Ganguly, Jayanta; Ghosh, Manas

2015-01-01

We make a rigorous exploration of the profiles of off-diagonal components of frequency-dependent linear (α xy , α yx ), first nonlinear (β xyy , β yxx ), and second nonlinear (γ xxyy , γ yyxx ) polarizabilities of quantum dots driven by Gaussian white noise. The quantum dot is doped with repulsive Gaussian impurity. Noise has been applied additively and multiplicatively to the system. An external oscillatory electric field has also been applied to the system. Gradual variations of external frequency, dopant location, and noise strength give rise to interesting features of polarizability components. The observations reveal intricate interplay between noise strength and dopant location which designs the polarizability profiles. Moreover, the mode of application of noise also modulates the polarizability components. Interestingly, in case of additive noise the noise strength has no role on polarizabilities whereas multiplicative noise invites greater delicacy in them. The said interplay provides a rather involved framework to attain stable, enhanced, and often maximized output of linear and nonlinear polarizabilities. - Highlights: • Linear and nonlinear polarizabilities of quantum dot are studied. • The polarizability components are off-diagonal and frequency-dependent. • Quantum dot is doped with a repulsive impurity. • Doped system is subject to Gaussian white noise. • Mode of noise application affects polarizabilities

Using the effect of alcohol as a comparison to illustrate the detrimental effects of noise on performance

Directory of Open Access Journals (Sweden)

Brett R.C Molesworth

2013-01-01

Full Text Available The aim of the present research is to provide a user-friendly index of the relative impairment associated with noise in the aircraft cabin. As such, the relative effect of noise, at a level typical of an aircraft cabin was compared with varying levels of alcohol intoxication in the same subjects. Since the detrimental effect of noise is more pronounced on non-native speakers, both native English and non-native English speakers featured in the study. Noise cancelling headphones were also tested as a simple countermeasure to mitigate the effect of noise on performance. A total of 32 participants, half of which were non-native English speakers, completed a cued recall task in two alcohol conditions (blood alcohol concentration 0.05 and 0.10 and two audio conditions (audio played through the speaker and noise cancelling headphones. The results revealed that aircraft noise at 65 dB (A negatively affected performance to a level comparable to alcohol intoxication of 0.10. The results also supported previous research that reflects positively on the benefits of noise cancelling headphones in reducing the effects of noise on performance especially for non-native English speakers. These findings provide for personnel involved in the aviation industry, a user-friendly index of the relative impairment associated with noise in the aircraft cabin as compared with the effects of alcohol. They also highlight the benefits of a simple countermeasure such as noise cancelling headphones in mitigating some of the detrimental effects of noise on performance.

Analysing Self Interference Cancellation in Full Duplex Radios

DEFF Research Database (Denmark)

Mahmood, Nurul Huda; Shafique Ansari, Imran; Berardinelli, Gilberto

2016-01-01

Full duplex communication promises a theoretical $100\\%$ throughput gain by doubling the number of simultaneous transmissions. Such compelling gains are conditioned on perfect cancellation of the self interference power resulting from simultaneous transmission and reception. Generally, self...... interference power is modelled as a noise-like constant level interference floor. However, experimental validations have shown that the self interference power is in practice a random variable depending on a number of factors such as the surrounding wireless environment and the degree of interference...... cancellation. In this study, we derive an analytical model for the residual self interference power, and demonstrate various applications of the derived model in analysing the performance of a Full Duplex radio. In general, full duplex communication is found to provide only modest throughput gains over half...

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

Partial Cancellation

Indian Academy of Sciences (India)

First page Back Continue Last page Overview Graphics. Partial Cancellation. Full Cancellation is desirable. But complexity requirements are enormous. 4000 tones, 100 Users billions of flops !!! Main Idea: Challenge: To determine which cross-talker to cancel on what “tone” for a given victim. Constraint: Total complexity is ...

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.

Intelligent coil drum with electromagnetic force cancellation for MRI equipment

International Nuclear Information System (INIS)

Qiu, Jinhao; Tani, Junji; Ohtomo, Kikuo

1995-01-01

High-level noise induced in MRI equipment has been a main obstacle for their wide application in medical diagnosis. The noise is caused by the vibration of the cylindrical shell used to support the gradient magnetic field coils in MRI. In this study, a reduced-size model of the coil drum is constructed using polymer films and piezoelectric films. The piezoelectric films are divided into several patches and two patches of them are used as integrated bending actuators in the control of the shell vibration so that no external actuators are needed. The disturbance cancellation method is used since the shell is excited by predetermined forces. The relationships between the size, position and output magnitude of the actuators and control effect are also discussed. Experiment and simulation were carried out for the cases where the shell was excited at resonance frequencies of three main modes. The results show that the disturbance cancellation method in combination with the piezoelectric film actuators can successfully suppress the forced vibration of the shell and that the minimum amplitude of controlled vibration varies with the resonance mode. (author)

A New Probe Noise Approach For Acoustic Feedback Cancellation In Hearing Aids

DEFF Research Database (Denmark)

Guo, Meng; Jensen, Søren Holdt; Jensen, Jesper

Acoustic feedback is a big challenge in hearing aids. If not appropriately treated, the feedback limits the maximum possible amplification and may lead to significant sound distortions. In a state-of-the-art hearing aid, an acoustic feedback cancellation (AFC) system is used to compensate...

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

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.

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.

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

Science.gov (United States)

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

2011-01-01

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

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

Directory of Open Access Journals (Sweden)

Shang-Yin Lee

2011-01-01

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

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.

Reduction of airfoil trailing edge noise by trailing edge blowing

International Nuclear Information System (INIS)

Gerhard, T; Carolus, T; Erbslöh, S

2014-01-01

The paper deals with airfoil trailing edge noise and its reduction by trailing edge blowing. A Somers S834 airfoil section which originally was designed for small wind turbines is investigated. To mimic realistic Reynolds numbers the boundary layer is tripped on pressure and suction side. The chordwise position of the blowing slot is varied. The acoustic sources, i.e. the unsteady flow quantities in the turbulent boundary layer in the vicinity of the trailing edge, are quantified for the airfoil without and with trailing edge blowing by means of a large eddy simulation and complementary measurements. Eventually the far field airfoil noise is measured by a two-microphone filtering and correlation and a 40 microphone array technique. Both, LES-prediction and measurements showed that a suitable blowing jet on the airfoil suction side is able to reduce significantly the turbulence intensity and the induced surface pressure fluctuations in the trailing edge region. As a consequence, trailing edge noise associated with a spectral hump around 500 Hz could be reduced by 3 dB. For that a jet velocity of 50% of the free field velocity was sufficient. The most favourable slot position was at 90% chord length

Wind noise within and across behind-the-ear and miniature behind-the-ear hearing aids.

Science.gov (United States)

Zakis, Justin A; Hawkins, Daniel J

2015-10-01

Previous studies investigated wind noise with Behind-The-Ear (BTE) hearing aids, but not the more common mini-BTE style of device, which typically has a smaller shell and microphones located more deeply behind the pinna. The current study investigated wind-noise levels across one BTE and two mini-BTE devices, and between the front and rear omni-directional microphones within devices. Levels were measured at two wind speeds (3 and 6 m/s) and 36 wind azimuths (10° increments). The pattern of wind-noise level versus azimuth was similar across mini-BTE devices, and differed for the BTE device. However, mean levels were markedly different across mini-BTE devices, and could be higher, lower, or similar to those of the BTE device. For within-device level differences, the pattern and mean across azimuth were similar across mini-BTE devices, and differed for the BTE device. Wind noise had the potential to slightly or severely reduce speech intelligibility at 3 or 6 m/s, respectively, across all devices.

Preamplifier with ultra low frequency cutoff for infrasonic condenser microphone

DEFF Research Database (Denmark)

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

2012-01-01

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

Optical microphone

Energy Technology Data Exchange (ETDEWEB)

Veligdan, J.T.

2000-01-11

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

Quiet comfort: noise, otherness, and the mobile production of personal space.

Science.gov (United States)

Hagood, Mack

2011-01-01

Marketing, news reports, and reviews of Bose QuietComfort noise-canceling headphones position them as essential gear for the mobile rational actor of the neoliberal market—the business traveler. This article concerns noise-canceling headphones’ utility as soundscaping devices, which render a sense of personal space by mediating sound. The airplane and airport are paradoxical spaces in which the pursuit of freedom impedes its own enjoyment. Rather than fight the discomforts of air travel as a systemic problem, travelers use the tactic of soundscaping to suppress the perceived presence of others. Attention to soundscaping enables the scholar to explore relationships between media, space, freedom, otherness, and selfhood in an era characterized by neoliberalism and increased mobility. Air travel is a moment in which people with diverse backgrounds, beliefs, and bodies crowd together in unusually close proximity. Noise is the sound of individualism and difference in conflict. Noise is othered sound, and like any type of othering, the perception of noise is socially constructed and situated in hierarchies of race, class, age, and gender. The normative QuietComfort user in media representations is white, male, rational, monied, and mobile; women, children, and “chatty” passengers are cast as noisemakers. Moreover, in putting on noise-canceling headphones, diverse selves put on the historically Western subjectivity that has been built into their technology, one that suppresses the noise of difference in favor of the smooth circulation of people, information, and commodities.

Electrocardiogram de-noising based on forward wavelet transform ...

Indian Academy of Sciences (India)

Ratio (SNR) and Mean Square Error (MSE) computations showed that our proposed ... This technique permits to cancel noises and retain the informa- tion of the ... Wavelet analysis is used for transforming the signal under investigation into joined temporal and ... introduced the BWT in our proposed ECG de-noising system.

Design and analysis of diaphragms in dynamic microphones

Directory of Open Access Journals (Sweden)

Zi-Gui Huang

2015-07-01

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

Static pressure and temperature coefficients of working standard microphones

DEFF Research Database (Denmark)

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

2016-01-01

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

Dynamic Pressure Microphones

Science.gov (United States)

Werner, E.

In 1876, Alexander Graham Bell described his first telephone with a microphone using magnetic induction to convert the voice input into an electric output signal. The basic principle led to a variety of designs optimized for different needs, from hearing impaired users to singers or broadcast announcers. From the various sound pressure versions, only the moving coil design is still in mass production for speech and music application.

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.

Experimental characterization of vertical-axis wind turbine noise.

Science.gov (United States)

Pearson, C E; Graham, W R

2015-01-01

Vertical-axis wind turbines are wind-energy generators suitable for use in urban environments. Their associated noise thus needs to be characterized and understood. As a first step, this work investigates the relative importance of harmonic and broadband contributions via model-scale wind-tunnel experiments. Cross-spectra from a pair of flush-mounted wall microphones exhibit both components, but further analysis shows that the broadband dominates at frequencies corresponding to the audible range in full-scale operation. This observation has detrimental implications for noise-prediction reliability and hence also for acoustic design optimization.

An Integrated Real-Time Beamforming and Postfiltering System for Nonstationary Noise Environments

Directory of Open Access Journals (Sweden)

Gannot Sharon

2003-01-01

Full Text Available We present a novel approach for real-time multichannel speech enhancement in environments of nonstationary noise and time-varying acoustical transfer functions (ATFs. The proposed system integrates adaptive beamforming, ATF identification, soft signal detection, and multichannel postfiltering. The noise canceller branch of the beamformer and the ATF identification are adaptively updated online, based on hypothesis test results. The noise canceller is updated only during stationary noise frames, and the ATF identification is carried out only when desired source components have been detected. The hypothesis testing is based on the nonstationarity of the signals and the transient power ratio between the beamformer primary output and its reference noise signals. Following the beamforming and the hypothesis testing, estimates for the signal presence probability and for the noise power spectral density are derived. Subsequently, an optimal spectral gain function that minimizes the mean square error of the log-spectral amplitude (LSA is applied. Experimental results demonstrate the usefulness of the proposed system in nonstationary noise environments.

Integration of Bass Enhancement and Active Noise Control System in Automobile Cabin

Directory of Open Access Journals (Sweden)

Liang Wang

2008-01-01

Full Text Available With the advancement of digital signal processing technologies, consumers are more concerned with the quality of multimedia entertainment in automobiles. In order to meet this demand, an audio enhancement system is needed to improve bass reproduction and cancel engine noise in the cabins. This paper presents an integrated active noise control system that is based on frequency-sampling filters to track and extract the bass information from the audio signal, and a multifrequency active noise equalizer to tune the low-frequency engine harmonics to enhance the bass reproduction. In the noise cancellation mode, a maximum of 3 dB bass enhancement can be achieved with significant noise suppression, while higher bass enhancement can be achieved in the bass enhance mode. The results show that the proposed system is effective for solving both the bass audio reproduction and the noise control problems in automobile cabins.

Transverse emittance dilution due to coupler kicks in linear accelerators

Directory of Open Access Journals (Sweden)

Brandon Buckley

2007-11-01

Full Text Available One of the main concerns in the design of low emittance linear accelerators (linacs is the preservation of beam emittance. Here we discuss one possible source of emittance dilution, the coupler kick, due to transverse electromagnetic fields in the accelerating cavities of the linac caused by the power coupler geometry. In addition to emittance growth, the coupler kick also produces orbit distortions. It is common wisdom that emittance growth from coupler kicks can be strongly reduced by using two couplers per cavity mounted opposite each other or by having the couplers of successive cavities alternate from above to below the beam pipe so as to cancel each individual kick. While this is correct, including two couplers per cavity or alternating the coupler location requires large technical changes and increased cost for superconducting cryomodules where cryogenic pipes are arranged parallel to a string of several cavities. We therefore analyze consequences of alternate coupler placements. We show here that alternating the coupler location from above to below compensates the emittance growth as well as the orbit distortions. For sufficiently large Q values, alternating the coupler location from before to after the cavity leads to a cancellation of the orbit distortion but not of the emittance growth, whereas alternating the coupler location from before and above to behind and below the cavity cancels the emittance growth but not the orbit distortion. We show that cancellations hold for sufficiently large Q values. These compensations hold even when each cavity is individually detuned, e.g., by microphonics. Another effective method for reducing coupler kicks that is studied is the optimization of the phase of the coupler kick so as to minimize the effects on emittance from each coupler. This technique is independent of the coupler geometry but relies on operating on crest. A final technique studied is symmetrization of the cavity geometry in the

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.

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.

Advantages of directional hearing aid microphones related to room acoustics

NARCIS (Netherlands)

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

1991-01-01

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

Electromagnetic Investigation of a CMOS MEMS Inductive Microphone

Directory of Open Access Journals (Sweden)

Farès TOUNSI

2009-09-01

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

Improved Noise Minimum Statistics Estimation Algorithm for Using in a Speech-Passing Noise-Rejecting Headset

Directory of Open Access Journals (Sweden)

Seyedtabaee Saeed

2010-01-01

Full Text Available This paper deals with configuration of an algorithm to be used in a speech-passing angle grinder noise-canceling headset. Angle grinder noise is annoying and interrupts ordinary oral communication. Meaning that, low SNR noisy condition is ahead. Since variation in angle grinder working condition changes noise statistics, the noise will be nonstationary with possible jumps in its power. Studies are conducted for picking an appropriate algorithm. A modified version of the well-known spectral subtraction shows superior performance against alternate methods. Noise estimation is calculated through a multi-band fast adapting scheme. The algorithm is adapted very quickly to the non-stationary noise environment while inflecting minimum musical noise and speech distortion on the processed signal. Objective and subjective measures illustrating the performance of the proposed method are introduced.

Effective connectivity between superior temporal gyrus and Heschl's gyrus during white noise listening: linear versus non-linear models.

Science.gov (United States)

Hamid, Ka; Yusoff, An; Rahman, Mza; Mohamad, M; Hamid, Aia

2012-04-01

This fMRI study is about modelling the effective connectivity between Heschl's gyrus (HG) and the superior temporal gyrus (STG) in human primary auditory cortices. MATERIALS #ENTITYSTARTX00026; Ten healthy male participants were required to listen to white noise stimuli during functional magnetic resonance imaging (fMRI) scans. Statistical parametric mapping (SPM) was used to generate individual and group brain activation maps. For input region determination, two intrinsic connectivity models comprising bilateral HG and STG were constructed using dynamic causal modelling (DCM). The models were estimated and inferred using DCM while Bayesian Model Selection (BMS) for group studies was used for model comparison and selection. Based on the winning model, six linear and six non-linear causal models were derived and were again estimated, inferred, and compared to obtain a model that best represents the effective connectivity between HG and the STG, balancing accuracy and complexity. Group results indicated significant asymmetrical activation (p(uncorr) Model comparison results showed strong evidence of STG as the input centre. The winning model is preferred by 6 out of 10 participants. The results were supported by BMS results for group studies with the expected posterior probability, r = 0.7830 and exceedance probability, ϕ = 0.9823. One-sample t-tests performed on connection values obtained from the winning model indicated that the valid connections for the winning model are the unidirectional parallel connections from STG to bilateral HG (p model comparison between linear and non-linear models using BMS prefers non-linear connection (r = 0.9160, ϕ = 1.000) from which the connectivity between STG and the ipsi- and contralateral HG is gated by the activity in STG itself. We are able to demonstrate that the effective connectivity between HG and STG while listening to white noise for the respective participants can be explained by a non-linear dynamic causal model with

Discrete oscillator design linear, nonlinear, transient, and noise domains

CERN Document Server

Rhea, Randall W

2014-01-01

Oscillators are an essential part of all spread spectrum, RF, and wireless systems, and today's engineers in the field need to have a firm grasp on how they are designed. Presenting an easy-to-understand, unified view of the subject, this authoritative resource covers the practical design of high-frequency oscillators with lumped, distributed, dielectric and piezoelectric resonators. Including numerous examples, the book details important linear, nonlinear harmonic balance, transient and noise analysis techniques. Moreover, the book shows you how to apply these techniques to a wide range of os

Noise Reduction Efforts for Special Operations C-130 Aircraft Using Active Synchrophaser Control

National Research Council Canada - National Science Library

Hammond, Daryl; McKinley, Richard; Hale, Bill

1998-01-01

.... A more complicated approach uses an active noise cancellation (ANC) system, which offers improved performance that can augment passive methods to significantly reduce both internal and external aircraft noise...

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...

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

Science.gov (United States)

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

2017-05-01

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

Robustness of quantum correlations against linear noise

International Nuclear Information System (INIS)

Guo, Zhihua; Cao, Huaixin; Qu, Shixian

2016-01-01

Relative robustness of quantum correlations (RRoQC) of a bipartite state is firstly introduced relative to a classically correlated state. Robustness of quantum correlations (RoQC) of a bipartite state is then defined as the minimum of RRoQC of the state relative to all classically correlated ones. It is proved that as a function on quantum states, RoQC is nonnegative, lower semi-continuous and neither convex nor concave; especially, it is zero if and only if the state is classically correlated. Thus, RoQC not only quantifies the endurance of quantum correlations of a state against linear noise, but also can be used to distinguish between quantum and classically correlated states. Furthermore, the effects of local quantum channels on the robustness are explored and characterized. (paper)

Noise reduction in binaural hearing aids : Analyzing the benefit over monaural systems

NARCIS (Netherlands)

Srinivasan, S.

2008-01-01

Binaural beamforming using signals from both left and right hearing aids offers greater potential for noise reduction than using signals from a single aid. However, wireless transmission of data imposes power constraints. Since most modern hearing aids have multiple microphones and are capable of

Performance analysis of power-efficient adaptive interference cancelation in fading channels

KAUST Repository

Radaydeh, Redha Mahmoud Mesleh; Alouini, Mohamed-Slim

2010-01-01

This paper analyzes the performance of a -steering scheme for highly correlated receive antennas in the presence of statistically unordered co-channel interferers over multipath fading channels. An adaptive activation of receive antennas according to the interfering signals fading conditions is considered in the analysis. Analytical expressions for various system performance measures, including the outage probability, average error probability of different signaling schemes, and raw moments of the combined signal-to-interference-plus-noise ratio (SINR) are obtained in exact forms. Numerical and simulation results for the performance-complexity tradeoff of this scheme is presented and then compared with that of full-size arbitrary interference cancelation and no cancelation scenarios. ©2010 IEEE.

Performance analysis of power-efficient adaptive interference cancelation in fading channels

KAUST Repository

Radaydeh, Redha Mahmoud Mesleh

2010-12-01

This paper analyzes the performance of a -steering scheme for highly correlated receive antennas in the presence of statistically unordered co-channel interferers over multipath fading channels. An adaptive activation of receive antennas according to the interfering signals fading conditions is considered in the analysis. Analytical expressions for various system performance measures, including the outage probability, average error probability of different signaling schemes, and raw moments of the combined signal-to-interference-plus-noise ratio (SINR) are obtained in exact forms. Numerical and simulation results for the performance-complexity tradeoff of this scheme is presented and then compared with that of full-size arbitrary interference cancelation and no cancelation scenarios. ©2010 IEEE.

A single antenna interference cancellation and adaptive technique based on ALOE

Directory of Open Access Journals (Sweden)

Xu Han

2017-02-01

Full Text Available A new type of single antenna interference cancellation (SAIC algorithm based on ALOE filtering module is introduced for co-channel interference cancellation in GSM/GPRS/EDGE downlink without changing the link structure of traditional receiver.Meanwhile,an adjacent frequency adaptive detection based on power spectrum estimation method is introduced to solve adjacent frequency interference and complex interference.Compared with traditional energy noise estimation method,the new method has simpler structure,less complexity,and can effectively improve the adaptability for various scenarios.The simulation results show that it can raise the resistance to co-channel frequency interference and adjacent frequency interference in multimode chips with low complexity,which improves the quality of 2G voice communication.

Fitness Effects of Network Non-Linearity Induced by Gene Expression Noise

Science.gov (United States)

Ray, Christian; Cooper, Tim; Balazsi, Gabor

2012-02-01

In the non-equilibrium dynamics of growing microbial cells, metabolic enzymes can create non-linearities in metabolite concentration because of non-linear degradation (utilization): an enzyme can saturate in the process of metabolite utilization. Increasing metabolite production past the saturation point then results in an ultrasensitive metabolite response. If the production rate of a metabolite depends on a second enzyme or other protein-mediated process, uncorrelated gene expression noise can thus cause transient metabolite concentration bursts. Such bursts are physiologically unnecessary and may represent a source of selection against the ultrasensitive switch, especially if the fluctuating metabolic intermediate is toxic. Selection may therefore favor correlated gene expression fluctuations for enzymes in the same pathway, such as by same-operon membership in bacteria. Using a modified experimental lac operon system, we are undertaking a combined theoretical-experimental approach to demonstrate that (i) the lac operon has an implicit ultrasensitive switch that we predict is avoided by gene expression correlations induced by same-operon membership; (ii) bacterial growth rates are sensitive to crossing the ultrasensitive threshold. Our results suggest that correlations in intrinsic gene expression noise are exploited by evolution to ameliorate the detrimental effects of nonlinearities in metabolite concentrations.

Linear energy divergences in Coulomb gauge QCD

OpenAIRE

Andrasi, A.

2011-01-01

The structure of linear energy divergences is analysed on the example of one graph to 3-loop order. Such dangerous divergences do cancel when all graphs are added, but next to leading divergences do not cancel out.

Reduction of Background Noise in the NASA Ames 40- by 80-Foot Wind Tunnel

Science.gov (United States)

Jaeger, Stephen M.; Allen, Christopher S.; Soderman, Paul T.; Olson, Larry E. (Technical Monitor)

1995-01-01

Background noise in both open-jet and closed wind tunnels adversely affects the signal-to-noise ratio of acoustic measurements. To measure the noise of increasingly quieter aircraft models, the background noise will have to be reduced by physical means or through signal processing. In a closed wind tunnel, such as the NASA Ames 40- by 80- Foot Wind Tunnel, the principle background noise sources can be classified as: (1) fan drive noise; (2) microphone self-noise; (3) aerodynamically induced noise from test-dependent hardware such as model struts and junctions; and (4) noise from the test section walls and vane set. This paper describes the steps taken to minimize the influence of each of these background noise sources in the 40 x 80.

Noise destroys the coexistence of periodic orbits of a piecewise linear map

International Nuclear Information System (INIS)

Wang Can-Jun; Yang Ke-Li; Qu Shi-Xian

2013-01-01

The effects of Gaussian white noise and Gaussian colored noise on the periodic orbits of period-5 (P-5) and period-6 (P-6) in their coexisting domain of a piecewise linear map are investigated numerically. The probability densities of some orbits are calculated. When the noise intensity is D = 0.0001, only the orbits of P-5 exist, and the coexisting phenomenon is destroyed. On the other hand, the self-correlation time τ of the colored noise also affects the coexisting phenomenon. When τ c c , only the orbits of P-5 appear, and the stability of the orbits of P-5 is enhanced. However, when τ > τ' c (τ c and τ c ' are critical values), only the orbits of P-6 exist, and the stability of the P-6 orbits is enhanced greatly. When τ c , the orbits of P-5 and P-6 coexist, but the stability of the P-5 orbits is enhanced and that of P-6 is weakened with τ increasing

Signal processing method for Johnson noise thermometry

International Nuclear Information System (INIS)

Hwang, I. G.; Moon, B. S.; Kinser, Rpger

2003-01-01

The development of Johnson Noise Thermometry requires a high sensitive preamplifier circuit to pick up the temperature-related noise on the sensing element. However, the random noise generated in this amplification circuit causes a significant erroneous influence to the measurement. This paper describes signal processing mechanism of the Johnson Noise Thermometry system which is underway of development in collaboration between KAERI and ORNL. It adopts two identical amplifier channels and utilizes a digital signal processing technique to remove the independent noise of each channel. The CPSD(Cross Power Spectral Density) function is used to cancel the independent noise and the differentiation of narrow or single frequency peak from the CPSD data separates the common mode electromagnetic interference noise

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...

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...

Reactor noise monitoring device

International Nuclear Information System (INIS)

Yamanaka, Hiroto.

1990-01-01

The present invention concerns a reactor noise monitoring device by detecting abnormal sounds in background noises. Vibration sounds detected by accelerometers are applied to a loose parts detector. The detector generates high alarm if there are sudden impact sounds in the background noises and applies output signals to an accumulation device. If there is slight impact sounds in the vicinity of any of the accelerometers, the accumulation device accumulates the abnormal sounds assumed to be generated from an identical site while synchronizing the waveforms for all of the channels. Then, the device outputs signals in which the background noises are cancelled, as detection signals. Therefore, S/N ratio can be improved and the abnormal sounds contained in the background noises can be detected, to thereby improve the accuracy for estimating the position where the abnormal sounds are generated. (I.S.)

The experimental plan of displacement- and frequency-noise free laser interferometer

International Nuclear Information System (INIS)

Kokeyama, K; Sato, S; Kawamura, S; Nishizawa, A; Chen, Y; Pai, A; Somiya, K; Ward, R; Sugamoto, A

2008-01-01

We present the partial demonstration of displacement- and laser-noise free interferometer (DFI) and the next experimental plan to examine the complete configuration. A part of the full implementation of DFI has been demonstrated to confirm the cancellation of beamsplitter displacements. The displacements were suppressed by about two orders of magnitude. The aim of the next experiment is to operate the system and to confirm the cancellation of all displacement noises, while the gravitational wave (GW) signals survive. The optical displacements will be simulated by electro-optic modulators (EOM). To simulate the GW contribution to laser lights, we will use multiple EOMs

Coupled influence of noise and damped propagation of impurity on linear and nonlinear polarizabilities of doped quantum dots

International Nuclear Information System (INIS)

Ganguly, Jayanta; Ghosh, Manas

2015-01-01

Highlights: • Linear and nonlinear polarizabilities of quantum dot are studied. • Quantum dot is doped with a repulsive impurity. • Doped system is subject to Gaussian white noise. • Dopant migrates under damped condition. • Noise-damping coupling affects polarizabilities. - Abstract: We investigate the profiles of diagonal components of static and frequency-dependent linear, first, and second nonlinear polarizabilities of repulsive impurity doped quantum dot. We have considered propagation of dopant within an environment that damps the motion. Simultaneous presence of noise inherent to the system has also been considered. The dopant has a Gaussian potential and noise considered is a Gaussian white noise. The doped system is exposed to an external electric field which could be static or time-dependent. Noise undergoes direct coupling with damping and the noise-damping coupling strength appears to be a crucial parameter that designs the profiles of polarizability components. This happens because the coupling strength modulates the dispersive and asymmetric character of the system. The frequency of external field brings about additional features in the profiles of polarizability components. The present investigation highlights some useful features in the optical properties of doped quantum dots

A 60-dB linear VGA with novel exponential gain approximation

International Nuclear Information System (INIS)

Zhou Jiaye; Tan Xi; Wang Junyu; Tang Zhangwen; Min Hao

2009-01-01

A CMOS variable gain amplifier (VGA) that adopts a novel exponential gain approximation is presented. No additional exponential gain control circuit is required in the proposed VGA used in a direct conversion receiver. A wide gain control voltage from 0.4 to 1.8 V and a high linearity performance are achieved. The three-stage VGA with automatic gain control (AGC) and DC offset cancellation (DCOC) is fabricated in a 0.18-μm CMOS technology and shows a linear gain range of more than 58-dB with a linearity error less than ±1 dB. The 3-dB bandwidth is over 8 MHz at all gain settings. The measured input-referred third intercept point (IIP3) of the proposed VGA varies from -18.1 to 13.5 dBm, and the measured noise figure varies from 27 to 65 dB at a frequency of 1 MHz. The dynamic range of the closed-loop AGC exceeds 56 dB, where the output signal-to-noise-and-distortion ratio (SNDR) reaches 20 dB. The whole circuit, occupying 0.3 mm 2 of chip area, dissipates less than 3.7 mA from a 1.8-V supply.

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

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.

Modular Engine Noise Component Prediction System (MCP) Program Users' Guide

Science.gov (United States)

Golub, Robert A. (Technical Monitor); Herkes, William H.; Reed, David H.

2004-01-01

This is a user's manual for Modular Engine Noise Component Prediction System (MCP). This computer code allows the user to predict turbofan engine noise estimates. The program is based on an empirical procedure that has evolved over many years at The Boeing Company. The data used to develop the procedure include both full-scale engine data and small-scale model data, and include testing done by Boeing, by the engine manufacturers, and by NASA. In order to generate a noise estimate, the user specifies the appropriate engine properties (including both geometry and performance parameters), the microphone locations, the atmospheric conditions, and certain data processing options. The version of the program described here allows the user to predict three components: inlet-radiated fan noise, aft-radiated fan noise, and jet noise. MCP predicts one-third octave band noise levels over the frequency range of 50 to 10,000 Hertz. It also calculates overall sound pressure levels and certain subjective noise metrics (e.g., perceived noise levels).

Robustness against parametric noise of nonideal holonomic gates

International Nuclear Information System (INIS)

Lupo, Cosmo; Aniello, Paolo; Napolitano, Mario; Florio, Giuseppe

2007-01-01

Holonomic gates for quantum computation are commonly considered to be robust against certain kinds of parametric noise, the cause of this robustness being the geometric character of the transformation achieved in the adiabatic limit. On the other hand, the effects of decoherence are expected to become more and more relevant when the adiabatic limit is approached. Starting from the system described by Florio et al. [Phys. Rev. A 73, 022327 (2006)], here we discuss the behavior of nonideal holonomic gates at finite operational time, i.e., long before the adiabatic limit is reached. We have considered several models of parametric noise and studied the robustness of finite-time gates. The results obtained suggest that the finite-time gates present some effects of cancellation of the perturbations introduced by the noise which mimic the geometrical cancellation effect of standard holonomic gates. Nevertheless, a careful analysis of the results leads to the conclusion that these effects are related to a dynamical instead of a geometrical feature

Robustness against parametric noise of nonideal holonomic gates

Science.gov (United States)

Lupo, Cosmo; Aniello, Paolo; Napolitano, Mario; Florio, Giuseppe

2007-07-01

Holonomic gates for quantum computation are commonly considered to be robust against certain kinds of parametric noise, the cause of this robustness being the geometric character of the transformation achieved in the adiabatic limit. On the other hand, the effects of decoherence are expected to become more and more relevant when the adiabatic limit is approached. Starting from the system described by Florio [Phys. Rev. A 73, 022327 (2006)], here we discuss the behavior of nonideal holonomic gates at finite operational time, i.e., long before the adiabatic limit is reached. We have considered several models of parametric noise and studied the robustness of finite-time gates. The results obtained suggest that the finite-time gates present some effects of cancellation of the perturbations introduced by the noise which mimic the geometrical cancellation effect of standard holonomic gates. Nevertheless, a careful analysis of the results leads to the conclusion that these effects are related to a dynamical instead of a geometrical feature.

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.

Improved Iterative Parallel Interference Cancellation Receiver for Future Wireless DS-CDMA Systems

Directory of Open Access Journals (Sweden)

Andrea Bernacchioni

2005-04-01

Full Text Available We present a new turbo multiuser detector for turbo-coded direct sequence code division multiple access (DS-CDMA systems. The proposed detector is based on the utilization of a parallel interference cancellation (PIC and a bank of turbo decoders. The PIC is broken up in order to perform interference cancellation after each constituent decoder of the turbo decoding scheme. Moreover, in the paper we propose a new enhanced algorithm that provides a more accurate estimation of the signal-to-noise-plus-interference-ratio used in the tentative decision device and in the MAP decoding algorithm. The performance of the proposed receiver is evaluated by means of computer simulations for medium to very high system loads, in AWGN and multipath fading channel, and compared to recently proposed interference cancellation-based iterative MUD, by taking into account the number of iterations and the complexity involved. We will see that the proposed receiver outperforms the others especially for highly loaded systems.

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

Directory of Open Access Journals (Sweden)

Christof Stieger

2017-08-01

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

A mean-field approach for an intercarrier interference canceller for OFDM

International Nuclear Information System (INIS)

Sakata, A; Kabashima, Y; Peleg, Y

2012-01-01

The similarity of the mathematical description of random-field spin systems to the orthogonal frequency-division multiplexing (OFDM) scheme for wireless communication is exploited in an intercarrier interference (ICI) canceller used in the demodulation of OFDM. The translational symmetry in the Fourier domain generically concentrates the major contribution of ICI from each subcarrier in the subcarrier’s neighbourhood. This observation in conjunction with the mean-field approach leads to the development of an ICI canceller whose necessary cost of computation scales linearly with respect to the number of subcarriers. It is also shown that the dynamics of the mean-field canceller are well captured by a discrete map of a single macroscopic variable, without taking the spatial and time correlations of estimated variables into account. (paper)

Combined Effect of Random Transmit Power Control and Inter-Path Interference Cancellation on DS-CDMA Packet Mobile Communications

Science.gov (United States)

Kudoh, Eisuke; Ito, Haruki; Wang, Zhisen; Adachi, Fumiyuki

In mobile communication systems, high speed packet data services are demanded. In the high speed data transmission, throughput degrades severely due to severe inter-path interference (IPI). Recently, we proposed a random transmit power control (TPC) to increase the uplink throughput of DS-CDMA packet mobile communications. In this paper, we apply IPI cancellation in addition to the random TPC. We derive the numerical expression of the received signal-to-interference plus noise power ratio (SINR) and introduce IPI cancellation factor. We also derive the numerical expression of system throughput when IPI is cancelled ideally to compare with the Monte Carlo numerically evaluated system throughput. Then we evaluate, by Monte-Carlo numerical computation method, the combined effect of random TPC and IPI cancellation on the uplink throughput of DS-CDMA packet mobile communications.

Infra-sound cancellation and mitigation in wind turbines

Science.gov (United States)

Boretti, Albert; Ordys, Andrew; Al Zubaidy, Sarim

2018-03-01

The infra-sound spectra recorded inside homes located even several kilometres far from wind turbine installations is characterized by large pressure fluctuation in the low frequency range. There is a significant body of literature suggesting inaudible sounds at low frequency are sensed by humans and affect the wellbeing through different mechanisms. These mechanisms include amplitude modulation of heard sounds, stimulating subconscious pathways, causing endolymphatic hydrops, and possibly potentiating noise-induced hearing loss. We suggest the study of infra-sound active cancellation and mitigation to address the low frequency noise issues. Loudspeakers generate pressure wave components of same amplitude and frequency but opposite phase of the recorded infra sound. They also produce pressure wave components within the audible range reducing the perception of the infra-sound to minimize the sensing of the residual infra sound.

Advantages of binaural amplification to acceptable noise level of directional hearing aid users.

Science.gov (United States)

Kim, Ja-Hee; Lee, Jae Hee; Lee, Ho-Ki

2014-06-01

The goal of the present study was to examine whether Acceptable Noise Levels (ANLs) would be lower (greater acceptance of noise) in binaural listening than in monaural listening condition and also whether meaningfulness of background speech noise would affect ANLs for directional microphone hearing aid users. In addition, any relationships between the individual binaural benefits on ANLs and the individuals' demographic information were investigated. Fourteen hearing aid users (mean age, 64 years) participated for experimental testing. For the ANL calculation, listeners' most comfortable listening levels and background noise level were measured. Using Korean ANL material, ANLs of all participants were evaluated under monaural and binaural amplification with a counterbalanced order. The ANLs were also compared across five types of competing speech noises, consisting of 1- through 8-talker background speech maskers. Seven young normal-hearing listeners (mean age, 27 years) participated for the same measurements as a pilot testing. The results demonstrated that directional hearing aid users accepted more noise (lower ANLs) with binaural amplification than with monaural amplification, regardless of the type of competing speech. When the background speech noise became more meaningful, hearing-impaired listeners accepted less amount of noise (higher ANLs), revealing that ANL is dependent on the intelligibility of the competing speech. The individuals' binaural advantages in ANLs were significantly greater for the listeners with longer experience of hearing aids, yet not related to their age or hearing thresholds. Binaural directional microphone processing allowed hearing aid users to accept a greater amount of background noise, which may in turn improve listeners' hearing aid success. Informational masking substantially influenced background noise acceptance. Given a significant association between ANLs and duration of hearing aid usage, ANL measurement can be useful for

Hearing aid noise suppression and working memory function

DEFF Research Database (Denmark)

Neher, Tobias; Wagener, Kirsten C.; Fischer, Rosa-Linde

2018-01-01

OBJECTIVE: Research findings concerning the relation between benefit from hearing aid (HA) noise suppression and working memory function are inconsistent. The current study thus investigated the effects of three noise suppression algorithms on auditory working memory and the relation with reading......-to-noise ratio (SNR) improvement. Auditory working memory was assessed at +6 dB SNR using listening span and N-back paradigms. STUDY SAMPLE: Twenty experienced HA users ages 55-80 with large differences in reading span. RESULTS: For the listening span measurements, there was an influence of HA setting...... on sentence-final word recognition and recall, with the directional microphones leading to ~6% better performance than the single-channel noise reduction. For the N-back measurements, there was substantial test-retest variability and no influence of HA setting. No interactions with reading span were found...

On the predictability of extreme events in records with linear and nonlinear long-range memory: Efficiency and noise robustness

Science.gov (United States)

Bogachev, Mikhail I.; Bunde, Armin

2011-06-01

We study the predictability of extreme events in records with linear and nonlinear long-range memory in the presence of additive white noise using two different approaches: (i) the precursory pattern recognition technique (PRT) that exploits solely the information about short-term precursors, and (ii) the return interval approach (RIA) that exploits long-range memory incorporated in the elapsed time after the last extreme event. We find that the PRT always performs better when only linear memory is present. In the presence of nonlinear memory, both methods demonstrate comparable efficiency in the absence of white noise. When additional white noise is present in the record (which is the case in most observational records), the efficiency of the PRT decreases monotonously with increasing noise level. In contrast, the RIA shows an abrupt transition between a phase of low level noise where the prediction is as good as in the absence of noise, and a phase of high level noise where the prediction becomes poor. In the phase of low and intermediate noise the RIA predicts considerably better than the PRT, which explains our recent findings in physiological and financial records.

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

Science.gov (United States)

Podboy, Gary G.; Horvath, Csaba

2010-01-01

A 48-microphone planar phased array system was used to acquire noise source localization data on a full-scale Williams International FJ44 turbofan engine. Data were acquired with the array at three different locations relative to the engine, two on the side and one in front of the engine. At the two side locations the planar microphone array was parallel to the engine centerline; at the front location the array was perpendicular to the engine centerline. At each of the three locations, data were acquired at eleven different engine operating conditions ranging from engine idle to maximum (take off) speed. Data obtained with the array off to the side of the engine were spatially filtered to separate the inlet and nozzle noise. Tones occurring in the inlet and nozzle spectra were traced to the low and high speed spools within the engine. The phased array data indicate that the Inflow Control Device (ICD) used during this test was not acoustically transparent; instead, some of the noise emanating from the inlet reflected off of the inlet lip of the ICD. This reflection is a source of error for far field noise measurements made during the test. The data also indicate that a total temperature rake in the inlet of the engine is a source of fan noise.

Canceling effect: a natural mechanism to reduce the effects of global warming

Science.gov (United States)

Razavi, Bahar S.; Blagodatskaya, Evgenia; kuzyakov, Yakov

2016-04-01

The temperature sensitivity of enzymes responsible for organic matter decomposition in soil is crucial for predicting the effects of global warming on the carbon cycle and sequestration. We tested the hypothesis that differences in temperature sensitivity of enzyme kinetic parameters Vmax and Km will lead to a canceling effect: strong reduction of temperature response of catalytic reactions. Short-term temperature response of Vmax and Km of three hydrolytic enzymes responsible for decomposition of cellulose (β-glucosidase, cellobiohydrolase) and hemicelluloses (xylanase) were analyzed in situ from 0 to 40 °C. The apparent activation energy varied between enzymes from 20.7 to 35.2 kJ mol-1 corresponding to the Q10 values of the enzyme activities of 1.4-1.9 (with Vmax-Q10 1.0-2.5 and Km-Q10 0.94-2.3). Temperature response of all tested enzymes fitted well to the Arrhenius equation. Despite that,the fitting of Arrhenius model revealed the non-linear increase of two cellulolytic enzymes activities with two distinct thresholds at 10-15 °C and 25-30 °C, which were less pronounced for xylanase. The nonlinearity between 10 and 15 °C was explained by 30-80% increase in Vmax. At 25-30 °C, however, the abrupt decrease of enzyme-substrate affinity was responsible for non-linear increase of enzyme activities. Our study is the first demonstrating nonlinear response of Vmax and Km to temperature causing canceling effect, which was most strongly pronounced at low substrate concentrations and at temperatures above 15 °C. Under cold climate, however, the regulation of hydrolytic activity by canceling in response to warming is negligible because canceling was never observed below 10 °C. The canceling, therefore, can be considered as natural mechanism reducing the effects of global warming on decomposition of soil organics at moderate temperatures. The non-linearity of enzyme responses to warming and the respective thresholds should therefore be investigated for other enzymes

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

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.

Stochastic multiresonance for a fractional linear oscillator with time-delayed kernel and quadratic noise

Science.gov (United States)

Guo, Feng; Wang, Xue-Yuan; Zhu, Cheng-Yin; Cheng, Xiao-Feng; Zhang, Zheng-Yu; Huang, Xu-Hui

2017-12-01

The stochastic resonance for a fractional oscillator with time-delayed kernel and quadratic trichotomous noise is investigated. Applying linear system theory and Laplace transform, the system output amplitude (SPA) for the fractional oscillator is obtained. It is found that the SPA is a periodical function of the kernel delayed-time. Stochastic multiplicative phenomenon appears on the SPA versus the driving frequency, versus the noise amplitude, and versus the fractional exponent. The non-monotonous dependence of the SPA on the system parameters is also discussed.

Commentary on the statistical properties of noise and its implication on general linear models in functional near-infrared spectroscopy.

Science.gov (United States)

Huppert, Theodore J

2016-01-01

Functional near-infrared spectroscopy (fNIRS) is a noninvasive neuroimaging technique that uses low levels of light to measure changes in cerebral blood oxygenation levels. In the majority of NIRS functional brain studies, analysis of this data is based on a statistical comparison of hemodynamic levels between a baseline and task or between multiple task conditions by means of a linear regression model: the so-called general linear model. Although these methods are similar to their implementation in other fields, particularly for functional magnetic resonance imaging, the specific application of these methods in fNIRS research differs in several key ways related to the sources of noise and artifacts unique to fNIRS. In this brief communication, we discuss the application of linear regression models in fNIRS and the modifications needed to generalize these models in order to deal with structured (colored) noise due to systemic physiology and noise heteroscedasticity due to motion artifacts. The objective of this work is to present an overview of these noise properties in the context of the linear model as it applies to fNIRS data. This work is aimed at explaining these mathematical issues to the general fNIRS experimental researcher but is not intended to be a complete mathematical treatment of these concepts.