scispace - formally typeset
Search or ask a question

Showing papers on "Background noise published in 2008"


Journal ArticleDOI
TL;DR: In this paper, the influence of designed pulse sequences in restoring quantum coherence lost due to background noise in superconducting qubits was investigated, and it was shown that the qubit coherence time can be substantially enhanced by carefully engineered pulse sequences.
Abstract: We theoretically investigate the influence of designed pulse sequences in restoring quantum coherence lost due to background noise in superconducting qubits. We consider both $1/f$ noise and random telegraph noise and show that the qubit coherence time can be substantially enhanced by carefully engineered pulse sequences. Conversely, the time dependence of qubit coherence under external pulse sequences could be used as a spectroscopic tool for extracting the noise mechanisms in superconducting qubits, i.e., by using Uhrig's pulse sequence [Phys. Rev. Lett. 98, 100504 (2007)], one can obtain information about moments of the spectral density of noise. We also study the effect of pulse sequences on the evolution of the qubit affected by a strongly coupled fluctuator and show that the non-Gaussian features in decoherence are suppressed by the application of pulses.

455 citations


Journal ArticleDOI
TL;DR: The statistical distribution of background noise was analyzed for MR acquisitions with a single-channel and a 32-channel coil, with sum-of-squares (SoS) and spatial-matched-filter (SMF) data combination, with and without parallel imaging using k-space and image-domain algorithms, with real-part and conventional magnitude reconstruction and with several reconstruction filters.

221 citations


BookDOI
01 Jan 2008
TL;DR: In this paper, the authors present an approach for active noise control in a multi-channel audio system with the goal of reducing the amount of active noise in the system by using adaptive techniques.
Abstract: 1. Acoustical oceanography Models for Propagation Codes Transducer Arrays: structure, data acquisition, signal generation, calibration Sonar MFP Tomography Other Inverse Techniques Signal and Noise Characteristics 2. Active Noise Control Principles of adaptive techniques Plant modeling Sound/vibration field sensing Actuator characteristics and requirements Performance limitations Multi-channel systems Performance and complexity 3. Animal bioacoustics Recording and monitoring systems Models of echolocation Hearing performance and modelling Characteristics of calls Stimuli generation Locating and tracking Archives and Databases of signals 4. Architectural acoustics Room models Measurement of transmissions, absorption, reverberation, etc. Sound fields (definitions, criteria, measurement, typical values) MLS and other coded signals Auralization: Modelling techniques, listening modes, processing requirements, existing systems, performace Artificial reverberation Sound reinforcement Acoustic privacy 5. Audio engineering Transducer modeling Loudspeaker performance characteristics Audio recording and playback formats Audio-visual interaction ADC, DAC, and Codec technologies Multi-channel sound and Virtual audio Restoration Digital audio editing Effects generation 6. Auditory System, Hearing Modeling of hearing Thresholds and Masking Frequency and level discrimination Binaural hearing and spatialization HRTF HATS and other physical models Hearing aids Auditory illusions 7. Education in acoustics 8. Electroacoustics Microphone types and their characteristics Vibration sensors and their characteristics Acoustic actuators and their characteristics Smart sensors and actuators 9. Engineering acoustics 10. Infrasonics Background noise and source signals Sensors and their characteristics Propagation models Event detection Data archiving Source identification 11. Musical Acoustics Computer music synthesis and composition Computer music recognition and analysis Singing voice analysis, synthesis, and processing Instrument measurement, modeling and synthesis Coding and compression of music 12. Noise Noise source modeling Acoustic holography Atmospheric sound propagation Source localization Noise evaluation and Annoyance thresholds 13. Non-linear acoustics Propagation equations and codes Example non-linear systems Parametric array Measurement methods Detection of non-linearities 14. Psychoacoustics Perceptual models Cochlear implants Auditory alarms 15. Seismology Seismic Coda Acoustic Profiling Propagation modes and properties for modeling Seismo-acoustic coupling 16. Speech Characteristics of speech as signals Synthesis Recognition Intelligibility and quality metrics Corpus for tests Coding and compression Display and analysis 17. Strutural acoustics and vibration BEM, FEM, EA, etc. Actuator design and deployment Propagation and radiation Machine diagnostics and prognosis Modeling, measuring and analyzing shock Materials testing 18. Telecomm POTS Wideband Echo supression Hearing aids Handset, Headset, and Wireless standards Systems for handicapped users 19. Ultrasonics

185 citations


Journal ArticleDOI
TL;DR: Investigation of the impact of traffic noise on acoustic communication in a tree frog by way of an experimental approach using noise playback showed that in response to noise playback, males are not able to adjust their temporal or frequency call structures to increase efficiency of the information transfer.

151 citations


Journal ArticleDOI
TL;DR: This paper proposes a VSS-APA derived in the context of AEC that aims to recover the near-end signal within the error signal of the adaptive filter and is robust against near- end signal variations (including double-talk).
Abstract: The adaptive algorithms used for acoustic echo cancellation (AEC) have to provide (1) high convergence rates and good tracking capabilities, since the acoustic environments imply very long and time-variant echo paths, and (2) low misadjustment and robustness against background noise variations and double-talk. In this context, the affine projection algorithm (APA) and different versions of it are very attractive choices for AEC. However, an APA with a constant step-size parameter has to compromise between the performance criteria (1) and (2). Therefore, a variable step-size APA (VSS-APA) represents a more reliable solution. In this paper, we propose a VSS-APA derived in the context of AEC. Most of the APAs aim to cancel p (i.e., projection order) previous a posteriori errors at every step of the algorithm. The proposed VSS-APA aims to recover the near-end signal within the error signal of the adaptive filter. Consequently, it is robust against near-end signal variations (including double-talk). This algorithm does not require any a priori information about the acoustic environment, so that it is easy to control in practice. The simulation results indicate the good performance of the proposed algorithm as compared to other members of the APA family.

148 citations


Proceedings ArticleDOI
06 Jul 2008
TL;DR: Deterministic construction of projection matrices that provably guarantee reconstruction with high probability are developed and compared with some simple capacity lower and upper bounds and with the recently obtained capacity of the Gaussian erasure channel.
Abstract: We use recently developed convex programming techniques to reconstruct arbitrary sparse signals observed through projections onto a small-dimensional space in background noise in order to estimate and remove impulsive noise in an OFDM system. We develop deterministic construction of projection matrices that provably guarantee reconstruction with high probability. Finally, we compare the achievable rate using our novel method with some simple capacity lower and upper bounds and with the recently obtained capacity of the Gaussian erasure channel. For practical impulse probability the proposed scheme appears to be competitive. This scheme may find some application in DSL and powerline communications, where transmission is typically affected by intersymbol interference, Gaussian noise and impulsive noise.

139 citations


Patent
24 Apr 2008
TL;DR: In this paper, a grammar used in speech recognition for reliability in a plurality of operating environments having different background noise was tested using a mixed test speech utterance with recorded background noise.
Abstract: Methods, systems, and products for testing a grammar used in speech recognition for reliability in a plurality of operating environments having different background noise that include: receiving recorded background noise for each of the plurality of operating environments; generating a test speech utterance for recognition by a speech recognition engine using a grammar; mixing the test speech utterance with each recorded background noise, resulting in a plurality of mixed test speech utterances, each mixed test speech utterance having different background noise; performing, for each of the mixed test speech utterances, speech recognition using the grammar and the mixed test speech utterance, resulting in speech recognition results for each of the mixed test speech utterances; and evaluating, for each recorded background noise, speech recognition reliability of the grammar in dependence upon the speech recognition results for the mixed test speech utterance having that recorded background noise.

118 citations


Journal ArticleDOI
TL;DR: In this paper, the authors reviewed the literature to develop an understanding of the effects of noise and music on human performance and found that the effect of music on driving performance is quite similar to that of noise on task performance.
Abstract: The purpose of the present paper was to review the literature to develop an understanding of the effects of noise and music on human performance. The second purpose was to study the effects of music on a commonly performed task that is frequently accompanied by background music: driving. Background noise not only affects public health, but it also negatively affects human performance in such tasks as comprehension, attention, and vigilance. However, some studies have indicated that noise exposure may not affect simple vigilance. Despite music's distinct difference from noise it too affects human performance negatively and positively. The results are inconclusive on the effects of music and task performance. More specifically, the effects of music on driving performance are quite similar to that of noise on task performance. Music seems to alleviate driver stress and mild aggression while at times facilitating performance. However, during other conditions of music, driving performance is impaired. Different aspects of sound (i.e. volume, type, tempo) impact human performance differently. It is still unknown which aspect (music or noise) affects task performance to a greater degree.

111 citations


Journal ArticleDOI
TL;DR: A new stochastic ML DOA estimator is derived based on an iterative procedure which concentrates the log-likelihood function with respect to the signal and noise nuisance parameters in a stepwise fashion and a modified inverse iteration algorithm is presented for the estimation of the noise parameters.
Abstract: This correspondence investigates the direction-of-arrival (DOA) estimation of multiple narrowband sources in the presence of nonuniform white noise with an arbitrary diagonal covariance matrix. While both the deterministic and stochastic Cramer-Rao bound (CRB) and the deterministic maximum-likelihood (ML) DOA estimator under this model have been derived by Pesavento and Gershman, the stochastic ML DOA estimator under the same setting is still not available in the literature. In this correspondence, a new stochastic ML DOA estimator is derived. Its implementation is based on an iterative procedure which concentrates the log-likelihood function with respect to the signal and noise nuisance parameters in a stepwise fashion. A modified inverse iteration algorithm is also presented for the estimation of the noise parameters. Simulation results have shown that the proposed algorithm is able to provide significant performance improvement over the conventional uniform ML estimator in nonuniform noise environments and require only a few iterations to converge to the nonuniform stochastic CRB.

108 citations


Journal ArticleDOI
TL;DR: It was found that neurons track unexpectedly fast transients, as their response amplitude has no attenuation up to 200 Hz, higher than the limits set by passive membrane properties and average firing rate and is not affected by the rate of change of the input.
Abstract: Cortical neurons are often classified by current-frequency relationship. Such a static description is inadequate to interpret neuronal responses to time-varying stimuli. Theoretical studies suggested that single-cell dynamical response properties are necessary to interpret ensemble responses to fast input transients. Further, it was shown that input-noise linearizes and boosts the response bandwidth, and that the interplay between the barrage of noisy synaptic currents and the spike-initiation mechanisms determine the dynamical properties of the firing rate. To test these model predictions, we estimated the linear response properties of layer 5 pyramidal cells by injecting a superposition of a small-amplitude sinusoidal wave and a background noise. We characterized the evoked firing probability across many stimulation trials and a range of oscillation frequencies (1-1000 Hz), quantifying response amplitude and phase-shift while changing noise statistics. We found that neurons track unexpectedly fast transients, as their response amplitude has no attenuation up to 200 Hz. This cut-off frequency is higher than the limits set by passive membrane properties (approximately 50 Hz) and average firing rate (approximately 20 Hz) and is not affected by the rate of change of the input. Finally, above 200 Hz, the response amplitude decays as a power-law with an exponent that is independent of voltage fluctuations induced by the background noise.

107 citations


Journal ArticleDOI
TL;DR: Measurements of the F-22A Raptor during static engine run-ups show that significant nonlinear propagation effects occur for even intermediate-thrust engine conditions and at angles well away from the peak radiation angle, which suggests that these effects are likely to be common in the propagation of noise radiated by high-power aircraft.
Abstract: To address the question of the role of nonlinear effects in the propagation of noise radiated by high-power jet aircraft, extensive measurements were made of the F-22A Raptor during static engine run-ups. Data were acquired at low-, intermediate-, and high-thrust engine settings with microphones located 23–305m from the aircraft along several angles. Comparisons between the results of a generalized-Burgers-equation-based nonlinear propagation model and the measurements yield favorable agreement, whereas application of a linear propagation model results in spectral predictions that are much too low at high frequencies. The results and analysis show that significant nonlinear propagation effects occur for even intermediate-thrust engine conditions and at angles well away from the peak radiation angle. This suggests that these effects are likely to be common in the propagation of noise radiated by high-power aircraft.

Journal ArticleDOI
TL;DR: This correspondence presents a microphone array shape calibration procedure for diffuse noise environments by fitting the measured noise coherence with its theoretical model and then estimates the array geometry using classical multidimensional scaling.
Abstract: This correspondence presents a microphone array shape calibration procedure for diffuse noise environments. The procedure estimates intermicrophone distances by fitting the measured noise coherence with its theoretical model and then estimates the array geometry using classical multidimensional scaling. The technique is validated on noise recordings from two office environments.

Journal ArticleDOI
TL;DR: In this article, the transient dynamics of the Verhulst model perturbed by arbitrary non-Gaussian white noise is investigated based on the infinitely divisible distribution of the Levy process.
Abstract: The transient dynamics of the Verhulst model perturbed by arbitrary non-Gaussian white noise is investigated. Based on the infinitely divisible distribution of the Levy process we study the nonlinear relaxation of the population density for three cases of white non-Gaussian noise: (i) shot noise; (ii) noise with a probability density of increments expressed in terms of Gamma function; and (iii) Cauchy stable noise. We obtain exact results for the probability distribution of the population density in all cases, and for Cauchy stable noise the exact expression of the nonlinear relaxation time is derived. Moreover starting from an initial delta function distribution, we find a transition induced by the multiplicative Levy noise, from a trimodal probability distribution to a bimodal probability distribution in asymptotics. Finally we find a nonmonotonic behavior of the nonlinear relaxation time as a function of the Cauchy stable noise intensity.

Journal ArticleDOI
TL;DR: Larger noise effects on consonant identification emerged for L2 (Dutch) than L1 listeners, suggesting that task factors rather than L2 population differences underlie the results discrepancy.
Abstract: Speech recognition in noise is harder in second (L2) than first languages (L1). This could be because noise disrupts speech processing more in L2 than L1, or because L1 listeners recover better though disruption is equivalent. Two similar prior studies produced discrepant results: Equivalent noise effects for L1 and L2 (Dutch) listeners, versus larger effects for L2 (Spanish) than L1. To explain this, the latter experiment was presented to listeners from the former population. Larger noise effects on consonant identification emerged for L2 (Dutch) than L1 listeners, suggesting that task factors rather than L2 population differences underlie the results discrepancy.

PatentDOI
Pei Xiang1, Song Wang1, Kulkarni Prajakt1, Samir Kumar Gupta1, Choy Eddie L T1 
TL;DR: In this article, a multi-microphone active noise cancellation (MMANC) functionality is used to remove background noise from audio information picked up on microphones of the mobile audio device.
Abstract: A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.

Journal ArticleDOI
TL;DR: In this article, the role of colored noise on the neuron dynamics was investigated by the mean response time (MRT) of the neuron, and the authors found meaningful modifications of the resonant activation (RA) and noise enhanced stability (NES) phenomena due to the correlation time of the noise.
Abstract: We analyze the dynamics of the FitzHugh-Nagumo (FHN) model in the presence of colored noise and a periodic signal. Two cases are considered: (i) the dynamics of the membrane potential is affected by the noise, (ii) the slow dynamics of the recovery variable is subject to noise. We investigate the role of the colored noise on the neuron dynamics by the mean response time (MRT) of the neuron. We find meaningful modifications of the resonant activation (RA) and noise enhanced stability (NES) phenomena due to the correlation time of the noise. For strongly correlated noise we observe suppression of NES effect and persistence of RA phenomenon, with an efficiency enhancement of the neuronal response. Finally we show that the self-correlation of the colored noise causes a reduction of the effective noise intensity, which appears as a rescaling of the fluctuations affecting the FHN system.

Journal ArticleDOI
TL;DR: This work presents a feature extraction technique based on modeling temporal envelopes of the speech signal in narrow subbands using frequency domain linear prediction (FDLP), which provides an all-pole approximation of the Hilbert envelope of the signal obtained by linear prediction on cosine transform of the signals.
Abstract: Performance of a typical automatic speech recognition (ASR) system severely degrades when it encounters speech from reverberant environments. Part of the reason for this degradation is the feature extraction techniques that use analysis windows which are much shorter than typical room impulse responses. We present a feature extraction technique based on modeling temporal envelopes of the speech signal in narrow subbands using frequency domain linear prediction (FDLP). FDLP provides an all-pole approximation of the Hilbert envelope of the signal obtained by linear prediction on cosine transform of the signal. ASR experiments on speech data degraded with a number of room impulse responses (with varying degrees of distortion) show significant performance improvements for the proposed FDLP features when compared to other robust feature extraction techniques (average relative reduction of 24% in word error rate). Similar improvements are also obtained for far-field data which contain natural reverberation in background noise. These results are achieved without any noticeable degradation in performance for clean speech.

Journal ArticleDOI
TL;DR: Noise reduction together with a proper choice of features could improve the classification accuracy to 96.78%, making the automated analysis a possibility, and investigating the performance of a new segmentation algorithm based on pattern recognition.
Abstract: Obstructive sleep apnea (OSA) is a highly prevalent disease in which upper airways are collapsed during sleep, leading to serious consequences. Snoring is the earliest symptom of OSA, but its potential in clinical diagnosis is not fully recognized yet. The first task in the automatic analysis of snore-related sounds (SRS) is to segment the SRS data as accurately as possible into three main classes: snoring (voiced non-silence), breathing (unvoiced non-silence) and silence. SRS data are generally contaminated with background noise. In this paper, we present classification performance of a new segmentation algorithm based on pattern recognition. We considered four features derived from SRS to classify samples of SRS into three classes. The features—number of zero crossings, energy of the signal, normalized autocorrelation coefficient at 1 ms delay and the first predictor coefficient of linear predictive coding (LPC) analysis—in combination were able to achieve a classification accuracy of 90.74% in classifying a set of test data. We also investigated the performance of the algorithm when three commonly used noise reduction (NR) techniques in speech processing—amplitude spectral subtraction (ASS), power spectral subtraction (PSS) and short time spectral amplitude (STSA) estimation—are used for noise reduction. We found that noise reduction together with a proper choice of features could improve the classification accuracy to 96.78%, making the automated analysis a possibility.

Journal ArticleDOI
TL;DR: In order to eliminate excess noise in the operating room it may be necessary to adopt a multidisciplinary approach, including the implementation of effective standards, and the focusing of the surgical team on noise matters.
Abstract: This study is an evaluation of the problem of noise pollution in operating rooms. The high sound pressure level of noise in the operating theatre has a negative impact on communication between operating room personnel. The research took place at nine Greek public hospitals with more than 400 beds. The objective evaluation consisted of sound pressure level measurements in terms of L(eq), as well as peak sound pressure levels in recordings during 43 surgeries in order to identify sources of noise. The subjective evaluation consisted of a questionnaire answered by 684 operating room personnel. The views of operating room personnel were studied using Pearson's X(2) Test and Fisher's Exact Test (SPSS Version 10.00), a t-test comparison was made of mean sound pressure levels, and the relationship of measurement duration and sound pressure level was examined using linear regression analysis (SPSS Version 13.00). The sound pressure levels of noise per operation and the sources of noise varied. The maximum measured level of noise during the main procedure of an operation was measured at L(eq)=71.9 dB(A), L(1)=84.7 dB(A), L(10)=76.2 dB(A), and L(99)=56.7 dB(A). The hospital building, machinery, tools, and people in the operating room were the main noise factors. In order to eliminate excess noise in the operating room it may be necessary to adopt a multidisciplinary approach. An improvement in environment (background noise levels), the implementation of effective standards, and the focusing of the surgical team on noise matters are considered necessary changes.

Journal ArticleDOI
TL;DR: A novel postfilter is developed which suppresses late reverberation of the near-end speech, residual echo and background noise, and maintains a constant residual background noise level.
Abstract: Hands-free devices are often used in a noisy and reverberant environment. Therefore, the received microphone signal does not only contain the desired near-end speech signal but also interferences such as room reverberation that is caused by the near-end source, background noise and a far-end echo signal that results from the acoustic coupling between the loudspeaker and the microphone. These interferences degrade the fidelity and intelligibility of near-end speech. In the last two decades, post filters have been developed that can be used in conjunction with a single microphone acoustic echo canceller to enhance the near-end speech. In previous works, spectral enhancement techniques have been used to suppress residual echo and background noise for single microphone acoustic echo cancellers. However, dereverberation of the near-end speech was not addressed in this context. Recently, practically feasible spectral enhancement techniques to suppress reverberation have emerged. In this paper, we derive a novel spectral variance estimator for the late reverberation of the near-end speech. Residual echo will be present at the output of the acoustic echo canceller when the acoustic echo path cannot be completely modeled by the adaptive filter. A spectral variance estimator for the so-called late residual echo that results from the deficient length of the adaptive filter is derived. Both estimators are based on a statistical reverberation model. The model parameters depend on the reverberation time of the room, which can be obtained using the estimated acoustic echo path. A novel postfilter is developed which suppresses late reverberation of the near-end speech, residual echo and background noise, and maintains a constant residual background noise level. Experimental results demonstrate the beneficial use of the developed system for reducing reverberation, residual echo, and background noise.

Journal ArticleDOI
TL;DR: It was shown that noise reduction algorithms can have a large influence on localization and that the ADM only preserves localization in the forward direction over azimuths where limited or no noise reduction is obtained, and the statistical Wiener filter approach introduces a better combination of sound source localization and noise reduction performance than theADM approach.
Abstract: This paper evaluates the influence of three multimicrophone noise reduction algorithms on the ability to localize sound sources. Two recently developed noise reduction techniques for binaural hearing aids were evaluated, namely, the binaural multichannel Wiener filter (MWF) and the binaural multichannel Wiener filter with partial noise estimate (MWF-N), together with a dual-monaural adaptive directional microphone (ADM), which is a widely used noise reduction approach in commercial hearing aids. The influence of the different algorithms on perceived sound source localization and their noise reduction performance was evaluated. It is shown that noise reduction algorithms can have a large influence on localization and that (a) the ADM only preserves localization in the forward direction over azimuths where limited or no noise reduction is obtained; (b) the MWF preserves localization of the target speech component but may distort localization of the noise component. The latter is dependent on signal-to-noise ratio and masking effects; (c) the MWF-N enables correct localization of both the speech and the noise components; (d) the statistical Wiener filter approach introduces a better combination of sound source localization and noise reduction performance than the ADM approach.

Journal ArticleDOI
TL;DR: A new approach is presented to adapt the energy and spectral parameters of HMMs as well as their time derivatives to the modifications by the speech input in a reverberant environment to combine the adaptation to background noise and unknown frequency characteristics.

Patent
05 May 2008
TL;DR: In this paper, an earpiece (100) and a method (640) for acoustic management of multiple microphones is provided, which can include capturing an ambient acoustic signal from an Ambient Sound Microphone (ASM) to produce an electronic ambient signal, capturing in an ear canal an internal sound from an Ear Canal Microphone(ECM), measuring a background noise signal, and mixing the electronic ambient signals with the electronic internal signals in a ratio dependent on the background noise signals to produce a mixed signal.
Abstract: An earpiece (100) and a method (640) for acoustic management of multiple microphones is provided. The method can include capturing an ambient acoustic signal from an Ambient Sound Microphone (ASM) to produce an electronic ambient signal, capturing in an ear canal an internal sound from an Ear Canal Microphone (ECM) to produce an electronic internal signal, measuring a background noise signal, and mixing the electronic ambient signal with the electronic internal signal in a ratio dependent on the background noise signal to produce a mixed signal. The mixing can adjust an internal gain of the electronic internal signal and an external gain of the electronic ambient signal based on the background noise characteristics. The mixing can account for an acoustic attenuation level and an audio content level of the earpiece. Other embodiments are provided.

Journal ArticleDOI
TL;DR: This work proposes a new approach to the background subtraction method which operates in the colour space and manages the colour information in the segmentation process to detect and eliminate noise.

Journal ArticleDOI
TL;DR: This work proposes a novel approach where the noisy magnitude spectrum is recombined with a changed phase spectrum to produce a modified complex spectrum, which results in improved speech quality.
Abstract: Typical speech enhancement algorithms operate on the short-time magnitude spectrum, while keeping the short-time phase spectrum unchanged for synthesis. We propose a novel approach where the noisy magnitude spectrum is recombined with a changed phase spectrum to produce a modified complex spectrum. During synthesis, the low energy components of the modified complex spectrum cancel out more than the high energy components, thus reducing background noise. Using objective speech quality measures, informal subjective listening tests and spectrogram analysis, we show that the proposed method results in improved speech quality.

Journal ArticleDOI
TL;DR: High noise annoyance consistently correlated with frequent interference of activities and reducing noise at night (10 pm-7 am) was more important than during the rest of the day.
Abstract: This study evaluated road traffic noise annoyance in Canada in relation to activity interference, subject concerns about noise and self-reported distance to a major road. Random digit dialing was employed to survey a representative sample of 2565 Canadians 15years of age and older. Respondents highly annoyed by traffic noise were significantly more likely to perceive annoyance to negatively impact health, live closer to a heavily traveled road and report that traffic noise often interfered with daily activities. Sex, age, education level, community size and province had statistically significant associations with traffic noise annoyance. High noise annoyance consistently correlated with frequent interference of activities. Reducing noise at night (10pm–7am) was more important than during the rest of the day.

Journal ArticleDOI
TL;DR: It can be concluded that the background noise level is one of the important factors on the estimation of community annoyance from aircraft noise exposure.
Abstract: A study of community annoyance caused by exposures to civil aircraft noise was carried out in 20 sites around Gimpo and Gimhae international airports to investigate the effect of background noise in terms of dose-effect relationships between aircraft noise levels and annoyance responses under real conditions Aircraft noise levels were mainly measured using airport noise monitoring systems, B&K type 3597 Social surveys were administered to people living within 100 m of noise measurement sites The question relating to the annoyance of aircraft noise was answered on an 11-point numerical scale The randomly selected respondents, who were aged between 18 and 70 years, completed the questionnaire independently In total, 753 respondents participated in social surveys The result shows that annoyance responses in low background noise regions are much higher than those in high background noise regions, even though aircraft noise levels are the same It can be concluded that the background noise level is one of the important factors on the estimation of community annoyance from aircraft noise exposure

Journal ArticleDOI
TL;DR: A new radiated noise measurement technique is proposed in this paper with a twofold objective: to simplify the measurement procedure and to obtain more information about noise sources.
Abstract: One of the requirements that electronics circuits must satisfy comprises conducted and irradiated noise specifications. Whereas conducted noise is well covered in the literature, radiated noise is not. Radiated noise regulations impose limits on the noise measured 3 or 10 m away from electronic equipment. These measurements are usually made in anechoic rooms, which are very expensive. Moreover, the measurement procedure is not a ldquoplug-and-playrdquo feature, but requires a strict measuring protocol. Once the electronic circuit has been tested, the designer remains ignorant of the source of the problem should the regulation not be met. Hence, the procedure to make an electronic circuit comply with regulations is usually one of trial-and-error, in which the experience of the designer is essential. A new radiated noise measurement technique is proposed in this paper with a twofold objective: to simplify the measurement procedure and to obtain more information about noise sources. The main idea is to scan the electric/magnetic field at two arbitrary although known distances. From these measurements, the source reconstruction technique enables the identification of the noise sources in the surface of the circuit and the field estimation at any distance and the assessment of compliance with regulations. Moreover, if regulations are not met, the effect of modifying the noise source can be tested in order to ascertain how the circuit should be modified to comply with regulations.

Patent
27 Feb 2008
TL;DR: In this article, the presence or absence of objects is determined by interrogating or exciting transponders coupled to the objects using pulsed wide band frequency signals, where the presence/absence determination may take into account frequency and/or Q value to limit false detections.
Abstract: The presence or absence of objects is determined by interrogating or exciting transponders coupled to the objects using pulsed wide band frequency signals. Interrogation is broken down into a number of subsample scan cycles each having interrogation cycles a start time forward in time by a fraction of a period of an expected transponder response signal. Ambient or background noise is evaluated and a threshold adjusted based on the level of noise. Adjustment may be based on multiple noise measurements or samples. Noise detection may be limited, with emphasis placed on interrogation to increase the signal to noise ratio. Matched filtering may be employed. Presence/absence determination may take into account frequency and/or Q value to limit false detections. Appropriate acts may be taken if detected noise is out of defined limits of operation, for example shutting down interrogation and/or providing an appropriate indication.

Journal ArticleDOI
TL;DR: The results suggest that the default mode of brain function differs when assessed in the presence compared to the absence of scanner noise, with the presence of scanners noise perhaps adding attentional demands that diminish activation changes between rest and task in a nonlinear way within the default network.
Abstract: Studies have identified specific brain regions that increase activation during rest relative to attention-demanding tasks; these regions subserve the "default mode of brain function". Most of these studies have been conducted in the presence of scanner background noise (SBN). This noise has been shown to lead to altered attentional demands, and thus may modulate the default-mode network. Twelve subjects were examined during a rest condition that was contrasted with an auditory task. Words were presented either with SBN employing a conventional acquisition or without SBN using a sparse sampling approach. The number of experimental and resting trials was equated between the designs. Selecting the images in the condition with SBN that corresponded in time with the images in the condition without SBN made a direct comparison of the default-mode network (rest contrasted with active task) possible. There was typical activation of the default-mode network during rest versus task for both designs. However, SBN suppressed major components of the default-mode network, including medial prefrontal cortex, posterior cingulate, and precuneus. Our results suggest that the default mode of brain function differs when assessed in the presence compared to the absence of scanner noise, with the presence of scanner noise perhaps adding attentional demands that diminish activation changes between rest and task in a nonlinear way within the default network. Further studies are needed to clarify whether the use of a sparse sampling technique might enhance clinical utilities that have been proposed for analysis of the default-mode network.