Showing papers on "Noise published in 1993"

Accurate short-term analysis of the fundamental frequency and the harmonics-to-noise ratio of a sampled sound

Abstract: We present a straightforward and robust algorithm for periodicity detection, working in the lag (autocorrelation) domain. When it is tested for periodic signals and for signals with additive noise or jitter, it proves to be several orders of magnitude more accurate than the methods commonly used for speech analysis. This makes our method capable of measuring harmonics-to-noise ratios in the lag domain with an accuracy and reliability much greater than that of any of the usual frequency-domain methods. By definition, the best candidate for the acoustic pitch period of a sound can be found from the position of the maximum of the autocorrelation function of the sound, while the degree of periodicity (the harmonics-to-noise ratio) of the sound can be found from the relative height of this maximum. However, sampling and windowing cause problems in accurately determining the position and height of the maximum. These problems have led to inaccurate timedomain and cepstral methods for pitch detection, and to the exclusive use of frequency-domain methods for the determination of the harmonics-to-noise ratio. In this paper, I will tackle these problems. Table 1 shows the specifications of the resulting algorithm for two spectrally maximally different kinds of periodic sounds: a sine wave and a periodic pulse train; other periodic sounds give results between these. Table 1. The accuracy of the algorithm for a sampled sine wave and for a correctly sampled periodic pulse train, as a function of the number of periods that fit in the duration of a Hanning window. These results are valid for pitch frequencies up to 80% of the Nyquist frequency. These results were measured for a sampling frequency of 10 kHz and window lengths of 40 ms (for pitch) and 80 ms (for HNR), but generalize to other sampling frequencies and window lengths (see section 5).

Active noise control

Abstract: Active noise control exploits the long wavelengths associated with low frequency sound. It works on the principle of destructive interference between the sound fields generated by the original primary sound source and that due to other secondary sources, acoustic outputs of which can be controlled. The acoustic objectives of different active noise control systems and the electrical control methodologies that are used to achieve these objectives are examined. The importance of having a clear understanding of the principles behind both the acoustics and the electrical control in order to appreciate the advantages and limitations of active noise control is emphasized. A brief discussion of the physical basis of active sound control that concentrates on three-dimensional sound fields is presented. >

Low noise electronic system design

Abstract: From the Publisher: Emphasizes IC design concepts with additional support for discrete design where necessary. Describes noise sources and models; addresses practical problems of circuit design for low noise using negative feedback, filtering, component noise, measurement techniques and instrumentation; gives numerous examples of practical amplifier designs. Five chapters cover the use of SPICE and PSpice for low noise analysis and design.

Effect of personal and situational variables on noise annoyance in residential areas

Abstract: This study uses methods that control for noise level and data quality to objectively evaluate the evidence on 22 personal and situational explanations for annoyance with environmental noise in residential areas. The balance of the evidence from 464 findings drawn from 136 surveys suggests that annoyance is not affected to an important extent by ambient noise levels, the amount of time residents are at home, the type of interviewing method, or any of the nine demographic variables (age, sex, social status, income, education, home ownership, type of dwelling, length of residence, or receipt of benefits from the noise source). Annoyance is related to the amount of isolation from sound at home and to five attitudes (fear of danger from the noise source, noise prevention beliefs, general noise sensitivity, beliefs about the importance of the noise source, and annoyance with non‐noise impacts of the noise source). The evidence is too evenly divided to indicate whether changes in noise environments cause residents to be annoyed more, less, or about the same as would be expected in long‐established noise environments. The evidence shows that even at low noise levels (below DNL 55 dB), a small percentage are highly annoyed and that the extent of annoyance is related to noise exposure.

Noise, Noise Sensitivity and Psychiatric Disorder: Epidemiological and Psychophysiological Studies

Abstract: Noise, a prototypical environmental stressor, has clear health effects in causing hearing loss but other health effects are less evident. Noise exposure may lead to minor emotional symptoms but the evidence of elevated levels of aircraft noise leading to psychiatric hospital admissions and psychiatric disorder in the community is contradictory. Despite this there are well documented associations between noise exposure and changes in performance, sleep disturbance and emotional reactions such as annoyance. Moreover, annoyance is associated with both environmental noise level and psychological and physical symptoms, psychiatric disorder and use of health services. It seems likely that existing psychiatric disorder contributes to high levels of annoyance. However, there is also the possibility that tendency to annoyance may be a risk factor for psychiatric morbidity. Although noise level explains a significant proportion of the variance in annoyance, the other major factor, confirmed in many studies, is subjective sensitivity to noise. Noise sensitivity is also related to psychiatric disorder. The evidence for noise sensitivity being a risk factor for psychiatric disorder would be greater if it were a stable personality characteristic, and preceded psychiatric morbidity. The stability of noise sensitivity and whether it is merely secondary to psychiatric disorder or is a risk factor for psychiatric disorder as well as annoyance is examined in two studies in this monograph: a six-year follow-up of a group of highly noise sensitive and low noise sensitive women; and a longitudinal study of depressed patients and matched control subjects examining changes in noise sensitivity with recovery from depression. A further dimension of noise effects concerns the impact of noise on the autonomic nervous system. Most physiological responses to noise habituate rapidly but in some people physiological responses persist. It is not clear whether this sub-sample is also subjectively sensitive to noise and whether failure to habituate to environmental noise may also represent a biological indicator of vulnerability to psychiatric disorder. In these studies noise sensitivity was found to be moderately stable and associated with current psychiatric disorder and a disposition to negative affectivity. Noise sensitivity levels did fall with recovery from depression but still remained high, suggesting an underlying high level of noise sensitivity. Noise sensitivity was related to higher tonic skin conductance and heart rate and greater defence/startle responses during noise exposure in the laboratory. Noise sensitive people attend more to noises, discriminate more between noises, find noises more threatening and out of their control, and react to, and adapt to noises more slowly than less noise sensitive people.(ABSTRACT TRUNCATED AT 400 WORDS)

Random pulse width modulation techniques for converter fed drive systems-a review

Abstract: A review of the state of the art in RPWM (random pulse width modulation) theory and practice is presented. Topics covered include principles of RPWM, means of randomization, a review of the existing RPWM techniques, power spectra, implementation issues, and documented effects of RPWM on electric drive systems. A number of RWPM strategies have been reported, and a beneficial impact on acoustic noise and vibration has been unamiously agreed upon. Published studies have proven that significant improvement of the acoustic noise characteristics of the motors can be achieved at practically no extra expense in comparison with the systems with traditional, deterministic PWM strategies. >

Recognition of speech in additive and convolutional noise based on RASTA spectral processing

Abstract: RASTA (relative spectral) processing is studied in a spectral domain which is linear-like for small spectral values and logarithmic-like for large spectral values. Experiments with a recognizer trained on clean speech and test data degraded by both convolutional and additive noise show that doing RASTA processing in the new domain yields results comparable with those obtained by training the recognizer on known noise. >

On noise reduction methods for chaotic data.

Abstract: Recently proposed noise reduction methods for nonlinear chaotic time sequences with additive noise are analyzed and generalized. All these methods have in common that they work iteratively, and that in each step of the iteration the noise is suppressed by requiring locally linear relations among the delay coordinates, i.e., by moving the delay vectors towards some smooth manifold. The different methods can be compared unambiguously in the case of strictly hyperbolic systems corrupted by measurement noise of infinitesimally low level. It was found that all proposed methods converge in this ideal case, but not equally fast. Different problems arise if the system is not hyperbolic, and at higher noise levels. A new scheme which seems to avoid most of these problems is proposed and tested, and seems to give the best noise reduction so far. Moreover, large improvements are possible within the new scheme and the previous schemes if their parameters are not kept fixed during the iteration, and if corrections are included which take into account the curvature of the attracting manifold. Finally, the fact that comparison with simple low‐pass filters tends to overestimate the relative achievements of these nonlinear noise reduction schemes is stressed, and it is suggested that they should be compared to Wiener‐type filters.

A fourth-order bandpass sigma-delta modulator

Abstract: The modulator of a bandpass analog/digital (A/D) converter, with 63 dB signal/noise for broadcast AM bandwidth signals centered at 455 kHz, has been implemented by modifying a commercial digital-audio sigma-delta ( Sigma Delta ) converter. It is the first reported fully monolithic implementation of bandpass noise shaping and has applications to digital radio. >

Robust voice activity detection using cepstral features

Abstract: This paper reviews algorithms which rely on the analysis of time domain samples to provide energy and zero-crossing rates, together with more recent algorithms that use different methods for speech detection. We then examine a different approach using cepstral analysis, showing a high degree of amplitude and noise level independence. We show that a cepstral based algorithm exhibits a high degree of independence to levels of background noise and successful speech end-pointing can be achieved via thresholding cepstral distance measures. Through the use of a noise code-book we are able to provide a successful reference for Euclidean distance measures in the voice detection algorithm. >

Extremely simple nonlinear noise-reduction method.

Abstract: A very simple method to reduce noise in experimental data with nonlinear time evolution is presented. Locally constant fits are used to obtain a less noisy trajectory consistent with the dynamics as well as with the measured data. Neighborhoods are defined by coordinates both from the past and from the future. The method is applied to the H\'enon map and to a discretized form of the Mackey-Glass equation.

Communication over power lines

Abstract: The present invention describes an FSK carrier communication systems for transmitting and receiving data on a electric utility line even in the presence of extreme noise. The present invention uses a unique FSK transmit and receive protocol to distinguish between noise and actual data. By careful selection of the space and mark frequencies transmitted on the power line and by using a unique variable width data-bit protocol, the receiver can distinguish between actual data and corrupted information due to noise spikes with nearly error free communication.

Structure of Supersonic Jet Flow and Its Radiated Sound

Abstract: The present paper explores the use of large-eddy simulations as a tool for predicting noise from first principles. A high-order numerical scheme is used to perform large-eddy simulations of a supersonic jet flow with emphasis on capturing the time-dependent flow structure representating the sound source. The wavelike nature of this structure under random inflow disturbances is demonstrated. This wavelike structure is then enhanced by taking the inflow disturbances to be purely harmonic. Application of Lighthill's theory to calculate the far-field noise, with the sound source obtained from the calculated time-dependent near field, is demonstrated. Alternative approaches to coupling the near-field sound source to the far-field sound are discussed.

Speech recognition in noise by children with minimal degrees of sensorineural hearing loss.

Abstract: It is well recognized that the acoustical environment in a classroom is an important variable in the psychoeducational achievement of hearing-impaired children. To date, however, there remains a paucity of information concerning the importance of classroom acoustics for children with minimal

Adaptive noise equalization and recognition of microcalcification clusters in mammograms

Abstract: A statistical method is described for detection of microcalcifications in digital mammograms. It is shown that the detection performance depends strongly on a preprocessing step, in which the images are rescaled to equalize image noise. A robust algorithm is proposed for rescaling, which can be used to determine a proper scale conversion from a phantom recording. The same algorithm, however, can also be applied to the image to be processed itself. Such an adaptive approach, in which noise characteristics are estimated from the image at hand, appeared to be the basis for far better results than could be obtained by using a fixed scale conversion. The method used for detection is based on Bayesian techniques. A random field model is used to model spatial relations between the labels in an iterative segmentation process. Results of an experimental study using a set of 65 mammographic images digitized at 2048×2048 are presented.

Longitudinal Study of Hearing Aid Effectiveness. I: Objective Measures

Abstract: This report is the first of two detailing a longitudinal follow-up of hearing aid users. Sixty-five subjects were followed for 12 months post-hearing aid fitting. Objective tests included insertion gain, the Speech Perception in Noise (SPIN) test (Kalikow, Stevens & Elliot, 1977; Bilger, Neutzel, Rabinowitz, & Rzeczkowski, 1984) and the Nonsense Syllable Test (NST) (Levitt & Resnick, 1978) presented in quiet and noise backgrounds. Initially each subject's hearing aid was fit to the revised National Acoustic Laboratories prescriptive formula (NAL-R) (Byrne & Dillon, 1986) using insertion gain measures. Use gain, measured at 6 and 12 months post-fitting, indicated that subjects generally used those prescribed values, except for subjects in the steeply sloping configuration subgroup. The NST and SPIN tests were administered at the fitting and at 1, 3, 6, and 12 months post-fitting. No change in performance, or training effect, was found for the group or for factors of experience, degree of hearing loss, configuration of hearing loss, use time, or circuit type. Failure to demonstrate a training effect may be attributed, in part, to the fact that initial speech recognition testing was done with the hearing aid volume set at the prescribed values. None of the circuits used showed performance superiority, except when comparing scores for the NST obtained in a quiet background to those obtained in a background of speech-weighted noise. In that comparison, the users of adaptive filter circuits exhibited less deterioration of performance in a noise background.

Simulation studies of multiple dipole neuromagnetic source localization: model order and limits of source resolution

Abstract: Numerical simulation studies were performed using a multiple-dipole source model and a spherical approximation of the head to determine how the resolution of simultaneously active neuromagnetic sources depends on source modeling assumptions (i.e., number of assumed dipoles), actual source parameters (e.g., location, orientation, and moment), and measurement errors. Forward calculations were made for a series of source configurations in which the number of dipoles, specific dipole parameters, and noise levels were systematically varied. Simulated noisy field distributions were fit by multiple dipole models of increasing model order (1,2, . . ., 6), and alternative statistical approaches (i.e., percent of variance, reduced chi-square, and F-ratio) were compared for their effectiveness in determining adequate model order. Limits of spatial resolution were established for a variety of multisource configurations and noise conditions. Implications for the analysis of empirical data are discussed. >

Sigma-delta digital-to-analog converter with reduced noise

Abstract: A sigma-delta digital-to-analog converter (DAC) (40) receives oversampled input data representative of an analog signal. The data may be optionally interpolated to a higher rate in a interpolator (41). A noise-shaping sigma-delta modulator (42) is connected to the output of the interpolator (41). The output of the modulator (42) is provided to a finite impulse response (FIR) filter (43). The FIR filter (43) has a frequency response characteristic which reduces the shaped noise and aliased components. This noise has a tendency to intermodulate back into the DAC's passband. The FIR filter (43) uses a series of flip-flops (81, 82, 83) functioning as delay elements with well-controlled timing edges. The outputs of the flip-flops (81, 82, 83) control current sources (91, 92, 93) weighted according to corresponding filter coefficients. The outputs of the current sources (91, 92, 93) are then summed in a summing device such as an amplifier (101).

Acoustical and Environmental Robustness in Automatic Speech Recognition

Abstract: This dissertation describes a number of algorithms developed to increase the robustness of automatic speech recognition systems with respect to changes in the environment These algorithms attempt to improve the recognition accuracy of speech recognition systems when they are trained and tested in different acoustical environments, and when a desk-top microphone (rather than a close-talking microphone) is used for speech input Without such processing, mismatches between training and testing conditions produce an unacceptable degradation in recognition accuracy Two kinds of environmental variability are introduced by the use of desk-top microphones and different training and testing conditions: additive noise and spectral tilt introduced by linear filtering An important attribute of the novel compensation algorithms described in this thesis is that they provide joint rather than independent compensation for these two types of degradation Acoustical compensation is applied in our algorithms as an additive correction in the cepstral domain This allows a higher degree of integration within SPHINX, the Carnegie Mellon speech recognition system, that uses the cepstrum as its feature vector Therefore, these algorithms can be implemented very efficiently Processing in many of these algorithms is based on instantaneous signal-to-noise ratio (SNR), as the appropriate compensation represents a form of noise suppression at low SNRs and spectral equalization at high SNRs The compensation vectors for additive noise and spectral transformations are estimated by minimizing the differences between speech feature vectors obtained from a “standard” training corpus of speech and feature vectors that represent the current acoustical environment In our work this is accomplished by a minimizing the distortion of vector-quantized cepstra that are produced by the feature extraction module in SPHINX In this dissertation we describe several algorithms including the SNR-Dependent Cepstral Normalization, (SDCN) and the Codeword-Dependent Cepstral Normalization (CDCN) With CDCN, the accuracy of SPHINX when trained on speech recorded with a close-talking microphone and tested on speech recorded with a desk-top microphone is essentially the same obtained when the system is trained and tested on speech from the desk-top microphone An algorithm for frequency normalization has also been proposed in which the parameter of the bilinear transformation that is used by the signal-processing stage to produce frequency warping is adjusted for each new speaker and acoustical environment The optimum value of this parameter is again chosen to minimize the vector-quantization distortion between the standard environment and the current one In preliminary studies, use of this frequency normalization produced a moderate additional decrease in the observed error rate

Computer simulation of auditory stream segregation in alternating-tone sequences.

Abstract: A computer model is described that takes a novel approach to the problem of accounting for perceptual coherence in alternating pure-tone sequences by using simple physiological principles that operate at a low level. Using the same set of parameter values, the model is able to reproduce a number of phenomena associated with auditory stream segregation. These are (1) the buildup of stream segregation over time, (2) the temporal coherence and fission boundaries obtained from human listeners, and (3) the trill threshold. Whereas these phenomena are generally accounted for in terms of an auditory scene-analysis process that works on the basis of Gestalt perceptual principles, the operation of the model suggests that some Gestalt auditory grouping may be the product of low-level processes.

"An Eye for an Eye Leaves Everyone Blind": Cooperation and Accounting Systems

Abstract: Using computer simulations that permit degrees of cooperation and introduce "noise" into the environment, I explore the benefits of strategies in which actors use different accounting systems to track ongoing exchanges. By relaxing some stringent assumptions of past work, I chart the conditions under which cooperation may emerge when actors can show degrees of cooperation and when actors 'moves are misperceived. Results provide evidence that strategies employing a relaxed accounting system have many advantages.

Noise margin criteria for digital logic circuits

Abstract: Techniques for evaluating the noise margin for families of digital logic circuits are discussed and evaluated. It is shown that the technique of evaluating the -1 slope points on the inverter transfer function as used in most modern textbooks is not a valid and reliable approach to evaluating noise margin values. It is argued that the most reliable and reasonable criterion is to maximize the product of the two noise margins. This is equivalent to maximizing the area of a rectangle embedded within the loop formed by the transfer curves of an inverter pair. Most of the material presented can be found in the early literature on noise margin. However, because of the widespread use of the -1 slope criterion in modern textbooks, it is believed that a reexamination of basic approaches to noise margins is in order. >

Development of a directional hearing instrument based on array technology

Abstract: A directional hearing aid might be beneficial in reducing background noise in relation to the desired speech signal Conventional hearing aids with a directional cardioid microphone are insufficient because of the low directivity of cardioids Research was done to develop microphone(s) with strong directional characteristics using array techniques Particular emphasis was given to optimization and stability Free‐field simulations of several robust models show that a directivity index of 9 dB can be obtained at the higher frequencies Simulations were verified with a laboratory model The results of the measurements show a good agreement with the simulations Based on simulations and measurements, two portable models were developed and tested with a KEMAR manikin The KEMAR measurements show that the two models give an improvement of the signal‐to‐noise ratio of approximately 75 dB in a diffuse sound field It may be concluded that the developed microphones have the capability to reach a significant improvement of speech intelligibility in noise under practical circumstances

Electronic instrument for locating and diagnosing automotive chassis sounds

Abstract: A vibration and acoustic sound diagnostic instrument for use by a professional automotive maintenance mechanic which is capable of discriminating audible vibration sound and noise generated by under-chassis and under-hood parts and devices having mechanical faults. The instrument includes at least one acoustic vibration pick-up device adapted for mounting in contact with an automotive part or device for detecting and converting vibratory acoustic signals and sounds into electromagnetic signals and an electronics housing conformed to be held in the hand of the maintenance mechanic. Pre-amplifier circuitry within the housing is electrically coupled to the acoustic vibration pick-up device and to range selector circuitry for selecting sound level ranges respecting the electromagnetic signals. A decibel meter, mounted to the exterior of the housing, is electrically interconnected to the pre-amplifier for visually indicating changes and peaks in the sound levels detected by the transducer microphone. Operational audio amplifier circuitry within the housing is electrically interconnected to the pre-amplifier circuitry for converting the electromagnetic signals into secondary acoustic signals that may be listened to by the maintenance mechanic through an earphone headset electrically interconnected to the audio amplifier. The instrument is energized by a battery power supply contained within the housing.

A new method for on wafer noise measurement

Abstract: A method for measuring the noise parameters of MESFETs and HEMTs is presented. It is based on the fact that three independent noise parameters are sufficient to fully describe the device noise performance. It is shown that two noise parameters, R/sub n/ and mod Y/sub OPT/ mod , can be directly obtained from the frequency variation of the noise figure F/sub 50/ corresponding to a 50 Omega generator impedance. By using a theoretical relation between the intrinsic noise sources as additional data, the F/sub 50/ measurement only can provide the four noise parameters. A good agreement with more conventional techniques is obtained. >

A performance analysis of subspace-based methods in the presence of model error. II. Multidimensional algorithms

Abstract: For pt.I, see ibid., vol.40, no.7, p.1758-74 (1992). In pt.I the performance of the MUSIC algorithms for narrowband direction-of-arrival (DOA) estimation when the array manifold and noise covariance are not correctly modeled was investigated. This analysis is extended to multidimensional subspace-based algorithms including deterministic (or conditional) maximum likelihood, MD-MUSIC, weighted subspace fitting (WSF), MODE, and ESPRIT. A general expression for the variance of the DOA estimates that can be applied to any of the above algorithms and to any of a wide variety of scenarios is presented. Optimally weighted subspace fitting algorithms are presented for special cases involving random unstructured errors of the array manifold and noise covariance. It is shown that one-dimensional MUSIC outperforms all of the above multidimensional algorithms for random angle-independent array perturbations. >

A Simple Measure of Price Adjustment Coefficients

Abstract: One measure of market efficiency is the speed with which prices adjust to new information. We develop a simple approach to estimating these price adjustment coefficients by using the information in return processes. This approach is used to estimate the price adjustment coefficients for firms listed on the NYSE and the AMEX as well as for over-the-counter stocks. We find evidence of a lagged adjustment to new information in shorter return intervals for firms in all market value classes, and some support for the proposition that prices adjust much more slowly and with more noise for smaller firms.