Showing papers on "Background noise published in 1993"

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. >

Measurements and models of radio frequency impulsive noise for indoor wireless communications

Abstract: The authors present the results of average and impulsive noise measurements inside several office buildings and retail stores. The noise measurement system operated at 918 MHz, 2.44 GHz, and 4 GHz with a nominal 40-MHz, 3-dB RF bandwidth. Omnidirectional and directional antennas were used to investigate the characteristics and sources of RF noise in indoor channels. Statistical analyses of the measurements are presented in the form of peak amplitude probability distributions, pulse duration distributions, and interarrival time distributions. Simple first-order mathematical models for these statistical characterizations are also presented. These analyses indicate that photocopiers, printers (both line printers and cash register receipt printers), elevators, and microwave ovens are significant sources of impulse noise in office and retail environments. >

Maximum Permissible Ambient Noise Levels for Audiometric Test Rooms.

Abstract: This Standard speci es maximum permissible ambient noise levels (MPANLs) allowed in an audiometric test room that produce negligible masking ( 2 dB) of test signals presented at reference equivalent threshold levels speci ed in ANSI S3.6-1996 American National Standard Speci cation of Audiometers. The MPANLs are speci ed from 125 to 8000 Hz in octave and one-third octave band intervals for two audiometric testing conditions (ears covered and ears not covered) and for three test frequency ranges (125 to 8000 Hz, 250 to 8000 Hz, and 500 to 8000 Hz). The Standard is intended for use by all persons testing hearing and for distributors, installers, designers, and manufacturers of audiometric test rooms. This standard is a revision of ANSI S3.1-1991 American National Standard Maximum Permissible Ambient Noise Levels for Audiometric Test Rooms. ANSI S3.1-1999 (Revision of ANSI S3.1-1991) Reaffirmed by ANSI October 15, 2003 Reaffirmed by ANSI October 28, 2008 Reaffir med by ANSI on June 24, 2007 Reaffir med by ANSI on June 24, 2007 This is a preview of \"ANSI/ASA S3.1-1999 (...\". Click here to purchase the full version from the ANSI store.

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. >

Voice activity detector for speech signals in variable background noise

Abstract: A voice activity detector (VAD) which determines whether received voice signal samples contain speech by deriving parameters measuring short term time domain characteristics of the input signal, including the average signal level and the absolute value of any change in average signal level, and comparing the derived parameter values with corresponding thresholds, which are periodically monitored and updated to reflect changes in the level of background noise, thereby minimizing clipping and false alarms.

Optimal realizable filters and the minimum Euclidean distance principle. [for spatial light modulators]

Abstract: Minimizing a Euclidean distance in the complex plane optimizes a wide class of correlation metrics for filters implemented on realistic devices. The algorithm searches over no more than two real scalars (gain and phase). It unifies a variety of previous solutions for special cases (e.g., a maximum signal-to-noise ratio with colored noise and a real filter and a maximum correlation intensity with no noise and a coupled filter). It extends optimal partial information filter theory to arbitrary spatial light modulators (fully complex, coupled, discrete, finite contrast ratio, and so forth), additive input noise (white or colored), spatially nonuniform filter modulators, and additive correlation detection noise (including signaldependent noise). An appendix summarizes the algorithm as it is implemented in available computer code.

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

Voice activity detection for cellular networks

Abstract: Algorithms for voice activity detection in the presence of vehicular noise and babble noise are presented. The voice activity detector (VAD) designed for vehicular noise is an improvement upon the VAD adopted for the discontinuous transmission (DTX) mode of the GSM standard, and performs significantly better at low SNR levels. Work on a VAD that is suited for the babble noise environment is also briefly mentioned, and a scheme for combining the two VADs is proposed.

Optimal realizable filters and the minimum Euclidean distance principle.

Abstract: Minimizing a Euclidean distance in the complex plane optimizes a wide class of correlation metrics for filters implemented on realistic devices. The algorithm searches over no more than two real scalars (gain and phase). It unifies a variety of previous solutions for special cases (e.g., a maximum signal-to-noise ratio with colored noise and a real filter and a maximum correlation intensity with no noise and a coupled filter). It extends optimal partial information filter theory to arbitrary spatial light modulators (fully complex, coupled, discrete, finite contrast ratio, and so forth), additive input noise (white or colored), spatially nonuniform filter modulators, and additive correlation detection noise (including signal dependent noise).

Enhancement of speech coding in background noise for low-rate speech coder

Abstract: A speech coding system employs measurements of robust features of speech frames whose distribution are not strongly affected by noise/levels to make voicing decisions for input speech occurring in a noisy environment. Linear programing analysis of the robust features and respective weights are used to determine an optimum linear combination of these features. The input speech vectors are matched to a vocabulary of codewords in order to select the corresponding, optimally matching codeword. Adaptive vector quantization is used in which a vocabulary of words obtained in a quiet environment is updated based upon a noise estimate of a noisy environment in which the input speech occurs, and the "noisy" vocabulary is then searched for the best match with an input speech vector. The corresponding clean codeword index is then selected for transmission and for synthesis at the receiver end. The results are better spectral reproduction and significant intelligibility enhancement over prior coding approaches. Robust features found to allow robust voicing decisions include: low-band energy; zero-crossing counts adapted for noise level; AMDF ratio (speech periodicity) measure; low-pass filtered backward correlation; low-pass filtered forward correlation; inverse-filtered backward correlation; and inverse-filtered pitch prediction gain measure.

Industrial Noise Control: Fundamentals and Applications, Second Edition

Abstract: "The Nature and Measurement of Sound Physical Acoustics Levels and Spectra Character of Noise Sound Propagation Sound Measurement and Analysis Noise Control Methods Acoustical Materials Acoustical Enclosures Silencers, Mufflers, and Active Noise Control Reverberation Control Vibration Control Basic Sources of Noise: Character and Treatment Fans and Blowers Gas-Jet Noise Gear Noise Additional Topics and Case Histories Environmental Acoustics Sound Control in Buildings Community and Environmental Noise Community and Environmental Noise Regulations Personal Hearing Protection Appendixes Appendix A: The International System of Units Appendix B: Conversion Factors Appendix C: Standards and Procedures Appendix D: Recommended Descriptors and Abreviations Appendix E: Department of Labor Occupational Noise Exposure Standard and Permissible Ultrasonic Treshold Levels Appendix F: Sound Absorption Coefficients of Common Building Materials, Audiences, Seats, Musicians, etc. Appendix G: Noys as a Function of Sound Pressure Level, IP Appendix H: Regulations of Connecticut State Agencies Appendix I: Machines Enclosure Listing Catalog "

System for adaptively reducing noise in speech signals

Abstract: A method and system are provided for adaptively reducing noise in frames of digitized audio signals that may include both speech and background noise. Frames of digitized audio signals are processed to determine what attenuation (if any) should be applied to the current frame of digitized audio signals. Initially it is determined whether the current frame of digitized audio signals includes speech information, this determination being based upon an estimate of noise and on a speech threshold value. An attenuation value determined for the previous audio frame is modified based on this determination and applied to the current frame in order to minimize the background noise which thereby improves the quality of received speech. The attenuation applied to the audio frames is modified gradually on a frame-by-frame basis, each sample in a specific frame is attenuated using the value calculated for that frame. The adaptive noise reduction system may be advantageously applied to telecommunication systems in which portable radio transceivers communicate over RF channels because the adaptive noise reduction technique does not significantly increase data processing overhead.

Assessment of a directional microphone array for hearing‐impaired listeners

Abstract: Hearing‐impaired listeners often have great difficulty understanding speech in surroundings with background noise or reverberation. Based on array techniques, two microphone prototypes (broadside and endfire) have been developed with strongly directional characteristics [Soede et al., ‘‘Development of a new directional hearing instrument based on array technology,’’ J. Acoust. Soc. Am. 94, (1993)]. Physical measurements show that the arrays attenuate reverberant sound by 6 dB (free‐field) and can improve the signal‐to‐noise ratio by 7 dB in a diffuse noise field (measured with a KEMAR manikin). For the clinical assessment of these microphones an experimental setup was made in a sound‐insulated listening room with one loudspeaker in front of the listener simulating the partner in a discussion and eight loudspeakers placed on the edges of a cube producing a diffuse background noise. The hearing‐impaired subject wearing his own (familiar) hearing aid is placed in the center of the cube. The speech‐reception threshold in noise for simple Dutch sentences was determined with a normal single omnidirectional microphone and with one of the microphone arrays. The results of monaural listening tests with hearing impaired subjects show that in comparison with an omnidirectional hearing‐aid microphone the broadside and endfire microphone array gives a mean improvement of the speech reception threshold in noise of 7.0 dB (26 subjects) and 6.8 dB (27 subjects), respectively. Binaural listening with two endfire microphone arrays gives a binaural improvement which is comparable to the binaural improvement obtained by listening with two normal ears or two conventional hearing aids.

Noise attenuation system

Abstract: The invention relates to a noise attenuation system for voice signals. The invention is based on noise signal measurement during the tonal periods of the voice. The line spectrum structure of the spectrum of the tonal sounds is utilized in the noise measurement. In this way it is possible to measure the noise of different frequency bands with the aid of narrow-band filters.

Precision tracking with segmentation for imaging sensors

Abstract: Precision target tracking based on data obtained from imaging sensors when the target is not fully visible during tracking is addressed. The image is divided into several layers of gray level intensities and thresholded. A binary image is obtained and grouped into clusters using image segmentation. The association of the various clusters to the track to be estimated relies on both the motion and pattern recognition characteristics of the target. The centroid measurements of the clusters and the probabilistic data association filter (PDAF) are employed for state estimation. Expressions for the single-frame-based centroid measurement noise variance of the target cluster and the optimal parameters for cluster segmentation are given. Simulation results validate the expressions for the measurement noise variance as well as the performance predictions of the tracking method. For a dim synthetic target with strong background noise, subpixel accuracy in the range of 0.3-0.4 pixel RMS error with moderate

Speech detection device for the detection of speech end points based on variance of frequency band limited energy

Abstract: The device detects the beginning and ending portions of speech contained within an input signal based on the variance of frequency band limited energy within the signal. The use of the variance allows detection which is relatively independent of an absolute signal-to-noise ratio with the signal, and allows accurate detection within a wide variety of backgrounds such as music, motor noise, and background noise, such as other speakers. The device can be easily implemented using off-the-shelf hardware along with a high-speed special purpose digital signal processor integrated circuit.

Infrared intrusion sensor

Abstract: An infrared intrusion sensor comprising an array of infrared detectors (17), infrared collection optics which may include a Cassegrain-style telescope (11, 12), a focal plane scanning device (11, 12, 13), including dither means adapted to repetitively scan the infrared radiation across the detector array, signal process means and local or remote display means. The sensor may incorporate heterodyne detection techniques with a local oscillator signal derived from the scanning frequency of the focal plane scanning device. The sensor has a low false alarm rate and enhanced detection range. Also disclosed is a method of processing signals involving A/D conversion, integration of the digital signals to produce a background signal, phase sensitive detection of the digital signal producing a target signal and comparison of the background and target signals producing a difference signal. This difference signal is then integrated to produce a background noise signal and processed to become a threshold signal which is finally compared to the difference signal to produce an alarm signal.

Electronic Noise and Low Noise Design

Abstract: Random signals noise connected with layout or construction intrinsic noise noise circuit analysis noise models noise performance measurement computer modelling low-noise design. Appendices: constants noise model of linear two-part network noise descriptors di-pole fields.

Active noise reduction apparatus

Abstract: An apparatus for active reduction of noises transmitted from a noise source into a space. The active noise reduction apparatus comprises residual noise sensors for detecting the residual noises in the space. A reference signal is produced based upon the noise generating condition of the noise source. The reference signal is used, along with the detected residual noises, to drive control sound sources so as to reduce the noises in the space. A filter is adjusted to correspond to acoustic transfer characteristics between the control sound sources and the residual noise sensors. An identification sound is generated to correspond to the background noise level detected in the space and to the spectral distribution of the noises transmitted into the space. The coefficients of the filter are updated according to acoustic transfer characteristics between the control sound sources and the residual noise sensors. The acoustic transfer characteristics are obtained based upon the identification sound and the residual noises.

Acoustical properties of speech produced in noise presented through supra-aural earphones.

Abstract: This study determined the acoustical properties of speech known as Lombard Speech produced in background noise. Tape recordings were made for ten normally hearing adults (5 women, 5 men) reading connected speech (131 word passage "My Grandfather") at their most comfortable level in quiet and in wideband, traffic, and multitalker noise delivered through earphones at 70 and 90 dB SPL. Spectral analysis of the recordings revealed that, compared with speech in quiet, Lombard speech was characterized by: (1) an increase in overall SPL; (2) smaller vocal pitch shifts for female than male subjects; (3) shifts in spectral distributions of speech energy; and (4) the same spectral slope above 630 Hz regardless of subject gender, noise level, or noise type. Overall, the results of this study do not provide support for the theory that acoustical properties of Lombard speech are identical with loud speech produced in quiet.

A computationally efficient unified approach to the numerical analysis of the sensitivity and noise of semiconductor devices

Abstract: The authors present a computationally efficient unified approach to the numerical simulation of sensitivity and noise in majority-carrier semiconductor devices that is based on the extension to device simulation of the adjoint method for sensitivity and noise analysis of electrical networks. Sensitivity and device noise analysis based on physical models are shown to have a common background, since they amount to evaluating the small-signal device response to an impressed, distributed current source. This problem is addressed by means of a Green's function technique akin to Shockley's impedance field method. To allow the efficient numerical evaluation of the Green's function within the framework of a discretized physical model, inter-reciprocity concepts, based on the introduction of an adjoint device, are exploited. Examples of implementation involving GaAs MESFETs are discussed. >

Influence of background noise and microphone on the performance of the IBM Tangora speech recognition system

Abstract: With the intention of developing a robust speech recognizer largely immune to the vagaries of extrinsic changes, the authors investigated the effects of various background noises and microphones on the performance of the Tangora system. They identified several noisy locations such as the cafeteria and a secretary's office and included a relatively quiet office for comparison. They recorded isolated-word training and test data from one male and one female speaker at different locations employing several varieties of microphones. A typical experiment consisted of designing a speaker-independent HMM (hidden Markov model) system with one set of training data and decoding the test data collected at all locations. It was found that microphone characteristics had a significant impact on the robustness of the system. It was also observed that controlled contamination of the quiet training data with ambient noise improved the noise immunity of the recognizer, discounting the role of the Lombard effect in the studies. >

Hadamard spectroscopy with a two-dimensional detecting array.

Abstract: In a conventional grating spectrograph consisting of a single entrance slit, a grating, and a multichannel (imaging) detector, considerable light throughput advantage can be realized by replacement of the single entrance slit with a mask. This replacement can yield a signal-to-noise ratio increase because of increased light collection over an extended area of the mask when compared with a single slit. The mask produces a spectrum on the detector, which is the convolution of the mask pattern and the spectral distribution of the light source. To retrieve the spectrum, the spectrum has to be inverted. In special cases in which emission spectra are superimposed on weak backgrounds, the signal-to-noise advantage is preserved through the inversion process. Thus this technique is valuable in the observation of light sources that are produced by atomic or molecular emissions such as aurora, airglow, some interstellar emission, or laboratory spectra. Considerable signal-to-noise advantages can also be realized when the background noise of the imaging detector is not negligible. The spectral mixing of the light from the mask on the detector causes high photon fluxes on the detector, which tend to swamp the detector noise. This is a particularly important advantage in the application of CCD’s as detectors because they can have significant background noise. The technique was demonstrated by computer simulations and laboratory tests.

Sound reproduction system.

Abstract: The invention relates to a sound reproduction system comprising a source of a sound signal in electrical form, a first filter group (13) for dividing the signal derived from the sound signal source into several sound signals occurring in different frequency bands and an adjustable gain amplifier (2) for each frequency band for amplifying the sound signal occurring in said frequency band. To take into account the noise existing in the environment of the sound reproduction system in the adjustment of the system, the system further comprises means (3) for generating an electrical signal proportional to the background noise existing in the environment and means (CS1...CSN) for adjusting the gain of the amplifiers in response to said electrical signal proportional to the background noise.

Least mean square (LMS) normalizer for active sonar

Abstract: A normalizer based on a Least Mean Square (LMS) adaptive algorithm configured to provide effective normalization when the background noise is locally non-stationary and when the target may be subject to time spread of unknown extent. The LMS algorithm used in the normalizer includes an adaptive filter in both the primary and reference inputs as a means of adapting to variations in both the signal and noise statistics. The LMS algorithm is implemented on the logarithm of the data, so that the difference minimized in the LMS structure drives the ratio of the signal power to noise power to a constant value. The algorithm can be used as a range normalizer by running it over range in each doppler bin, or as a frequency normalizer by operating across doppler in each range bin. By continually adapting to the statistics present in the data, the normalizer more effectively deals with the variations in the noise and signal statistics.

Method of estimating delay in noise-affected voice channels

Abstract: The present invention relates to a method of reducing noise in a speech detection system. The phases of at least two noise-affected signals are estimated. The phase estimate and the phase compensation required for the noise reduction are performed in the frequency domain. The background noise and the transient behavior of the enclosed space are simultaneously estimated.