Showing papers on "Noise published in 1987"

FM communication system with improved response to rayleigh-faded received signals

Abstract: An improved FM communication system is disclosed in which noise bursts and pops in the audio output of the receiver, due to weak and Rayleigh-faded signals, are minimized by selectively attenuating the audio signal in the receiver. The amount and rate of attenuation for the receiver noise reduction circuits is determined by a control signal derived form either the average discriminator noise at frequencies above 3 kHz, or from a received RF signal strength indicator, or a combination of both. The audio output is attenuated during weak signals or Rayleigh fades only when the attenuation control signal increases above a predetermined signal-to-noise ratio threshold. Also described are further improvements to noise attenuation devices used in combination with the receiver noise reduction circuits to enhance audio quality in particular applications, e.g., FM repeater systems which may incorporate multiple transmitters in a simulcast configuration.

Intensity discrimination and loudness for tones in notched noise

Abstract: Fechner proposed that equally often noticed differences are subjectively equal. Parker and Schneider (1980) showed that a power-function representation for loudness was consistent with Fechner’s proposal for auditory intensity discrimination when 1-kHz tones were presented in a quiet background. In the present experiment, (1) intensity-increment thresholds were determined for 1-kHz tones in notched noise, and (2) the intensities of 1-kHz tones in quiet which produced the same loudnesses as the tones in notched noise were obtained from the same subjects. Intensityincrement thresholds in notched noise could not be accounted for by either the loudness level of the tones in notched noise or by the Parker and Schneider model. A new model is developed in which the effects of notched noise on discrimination performance are related to the effects of notched noise on loudness.

The Multiplex Disadvantage and Excess Low-Frequency Noise

Abstract: Analysis of the performance of a Fourier transform spectrometer with respect to source shot and flicker noises is presented. It was found that source shot noise uniformly distributes throughout the baseline, whereas source flicker noise remains localized about the generating spectral region(s).

Research note: Patterns of exposure of schoolchildren to amplified music

Abstract: The amplified-music listening habits and the hearing status of 1443 comprehensive schoolchildren have been surveyed by questionnaire In a stratified sample of them (n = 25), the levels at which they would listen to a personal cassette player (PCP) in various circumstances were measured The main conclusion drawn was reassuring: that these children seem to be more conservative in their music-listening habits than young people are commonly given credit for Nevertheless, music may sometimes make a small contribution to socioacusis and to the total noise dosage of those exposed to noise occupationally Some recommendations are made for manufacturers of PCPs, and for some restrictions on the use of PCPs in occupational noise

Speech recognition with a noise-adapting codebook

Abstract: Speech recognizers trained in quiet conditions but operating in noise usually have poor accuracy. This paper reports two methods for improving the accuracy of an LPC vector-quantization speech recognizer by adapting the vector codebook to noisy conditions. First, each codebook vector is changed to reflect the way people speak in noise. Second, the estimated spectrum of the background noise is added to the codebook vectors. These ideas have been tested on a total of 2400 utterances of digits recorded in a car by 4 speakers. A baseline word spotter similar to NTT's SPLIT system was modified by adapting its vector codebook to noise. This adapted codebook, when used with a new word decision criterion, yields error rates at least 4 times lower for noisy conditions. The accuracy is significantly better than without codebook adaptation techniques.

Adaptive noise reduction in aircraft communication systems

Abstract: In many military environments, such as fighter jet cockpits, the increasing use of digital communication systems has created a need for robust vocoders and speech recognition systems. However, the high level of ambient noise in such environments makes vocoders less intelligible and makes reliable speech recognition more difficult. One method of enhancing the noise-corrupted speech is adaptive noise cancellation. In previous research, this method was tested in a simulated cockpit environment, yielding impressive results. However, in new simulations, reflecting more realistic conditions, adaptive noise cancellation has been less successful. Spectral analysis of the data showed that the diffuseness of the ambient noise fieid, along With the microphone characteristics, has a significant effect on the performance of adaptive noise cancellation.

Evaluation of the Audible Distortion and Noise Produced by Digital Audio Converters

Abstract: Une methode de mesure revele les non-linearites du convertisseur numerique a faible niveau. Distorsion resultant de tels defauts

FM receiver having improved audio quality in response to Rayleigh faded received signals

Abstract: The present invention minimizes noise pops due to Rayleigh fades by controlling receiver gain relative to the average received signal strength so that only a predetermined level of limiting occurs prior to the recovery of the audio information. The amount of limiting is selected so that a significant Rayleigh fade will cause the amplified IF signal to drop below the limiting level and enter the linear amplification region. During the fade the magnitude of the signal provided to the discriminator decreases thereby decreasing the discriminator's output and minimizing the magnitude of the audio pop. Additional minimization of the subject noise pops is achieved by utilizing a discriminator which has a square law transfer characteristic so that the recovered audio level will decrease 2 dB for every 1 dB decrease in input signal level.

Natural flicker noise (“1/f noise”) in music

Abstract: A large class of musical selections exhibits a spectral density of audio power fluctuations characterized by a low-frequency behaviour typical of 1/f noise. We show that this 1/f behaviour follows from natural flicker noise theory.

Increased noise immunity in large vocabulary speech recognition with the aid of spectral subtraction

Abstract: This paper presents several ways of making the signal processing in the IBM speech recognition system more robust with respect to variations in the background noise level. The underlying problem is that the speech recognition system trains on the specific noise circumstances of the training session. A simple solution lays in the controlled addition of noise. The level of noise that has to be added in to effectively mask all background noise is rather high and causes a significant reduction in accuracy. Spectral subtraction does a better job in a limited number of cases, but the thresholding in spectral subtraction often leads to training problems in the hidden Markov model based recognition system. The best results were obtained by reintroducing a semi-natural background by adding noise after applying spectral subtraction.

Method for damping interfering (wind) noise in sound signals transmitted by hearing aids

Abstract: A method is proposed for damping interfering noise in the sound signal in the case of transmission by hearing aids. In this case, the sound signals detected are digitised, and the spectral distributions are determined by Fourier analysis. The temporal variations in the spectral distributions are monitored in a plurality of frequency windows. The respective temporal variations are compared with prescribed and, as the case may be, modified limit values. Depending on the comparison, control signals are produced which vary the amplitude of the spectral distributions and/or the cut-off frequencies in the respective frequency windows in such a way that the interference signals are suppressed. The changed spectral distributions are subsequently converted into synthetic sound signals which, for their part, are converted into acoustic signals.

Apparatus for detecting an electronic surveillance device

Abstract: A method and apparatus for detecting a covert audio transmitter in which a microphone which picks up audio frequencies in the room and a radio receiver tuned to an unidentified audio signal are compared to determine if a "bug" is in the room being searched. An innocuous type of noise is introduced into the area to be searched by having persons converse or by turning on a radio, and these audio frequency signals are picked up directly by a microphone associated with the bug. The signal received by the bug causes the bug's RF carrier to be modulated, and this is picked up by the bug detector's radio receiver. The received signal is demodulated and is compared with the audio signal picked up by the bug detector's microphone to determine if the demodulated radio signal corresponds to the sounds in the room. If the demodulated signal from the detector's radio receiver matches the audio information picked up by the detector's microphone, an indicator light is lit to inform the operator that a bug is present.

Enhancement and Masking of Loudness by Environmental Factors Vegetation and Noise

Abstract: The effect of sound, especially noise, on the behavior of people in various environments has been of long-standing interest to researchers. However, the reverse effect, that is, the influence of environmental features on the perception of sound, appears to have received comparatively little attention. One aspect of this problem is addressed in the present work, namely, the dependence of loudness on visually prominent characteristics of outdoor environments, such as vegetation. Three experiments were conducted to assess the effects of visibility of vegetation in outdoor sites on the loudness of neutral sounds (500 hz tones) ranging between 50 and 80 db. It was found that, when the ambient noise level was held approximately constant across sites, loudness increased as the percentage of visible vegetation increased. It is argued that expected noise levels are keyed to the extensiveness of visible vegetation and that discrepancies between actual and expected noise levels in outdoor environments result in enhancements of the loudness of the sounds heard there. (Author/TRRL)

Application of adaptive noise cancelling to diver voice communications

Abstract: This paper describes an adaptive noise cancellation technique suitable for diver voice communications in which the noise may be multisource. Such situations require close physical spacing between primary and noise reference microphones to maintain a high correlation between the noise picked up on the reference channel with that on the primary channel. Signal to noise ratio improvements predicted by computer simulation are presented and practical realisation problems are discussed.

Estimation of noise variance from the noisy AR signal and its application in speech enhancement

Abstract: In a number of applications involving the processing of noisy signals, it is desirable to know a priori the noise variance. We propose here a method of estimating the noise variance from the autoregressive (AR) signal corrupted by the additive white noise. This method first estimates the AR parameters from the high-order Yule-Walker equations and then uses these AR parameters to estimate the noise variance from the low-order Yule-Walker equations. The method is studied for a number of examples of noisy AR signals and its performance is found to be close to the Cramer-Rao lower bound for high signal-to-noise ratios. It is also used in a speech enhancement application where its performance is studied for stationary as well as nonstationary noise conditions. The results are found to be encouraging.

Binaural versus monaural loudness: Supersummation of tone partially masked by noise

Abstract: A series of three experiments used the method of magnitude estimation to examine binaural summation of the loudness of a 1000‐Hz tone heard in the quiet and against various backgrounds of masking noise. In the quiet, binaural loudness as measured in sones, is twice monaural loudness. Two conditions of noise masking acted to increase the ratio of binaural/monaural loudness in sones above 2:1—that is, to produce supersummation. (1) When tone was presented to both ears, but masking noise to just one ear (dichotic stimulation), the loudness of the binaural tone was 30%–35% greater than the sum of the loudnesses of the monaural components. This increase in summation provides a suprathreshold analog to increases in threshold sensitivity observed with dichotic stimulation (masking‐level differences). (2) Supersummation was also evident when tone and noise alike were presented to both ears (diotic stimulation); here, the binaural tone’s loudness was 10%–25% greater than the sum of the monaural components. The inc...

Effect of noise on spectral library searching performance

Abstract: Modele empirique de profile de bruit dans une methode couplee chromatographie en phase gazeuse, spectrometrie IR a transformee de Fourier. Influence de spectres experimentaux a faible rapport signal/bruit sur l'identification des composes

Factors Related To Spectral Subtraction For Speech In Noise Enhancement

Abstract: In this paper results are presented concerning the use of spectral subtraction for enhancing speech corrupted by additive noise. In this study several variations of the spectral subtraction process have been implemented and tested. The overall results provide insight into the use of spectral subtraction for speech enhancement. As a result of this study, several conclusions are presented regarding spectral subtraction, its effects on noise corrupted speech, and its ability to enhance speech intelligibility and listenability. Finally, the use of spectral subtraction as a preprocessor for speech feature extraction is considered.

Effects of Noise and Hearing Loss upon Frequency Discrimination

Abstract: Frequency discrimination in quiet and in the presence of several levels of low-frequency masking noise was measured at 500, 1 000 and 2 000 Hz in both normal-hearing and hearing-impaired subjects. The test signals were presented at 80 dB SPL; all test signals were at least 10 dB above each subject's masked threshold. As a group, the hearing impaired subjects had larger frequency difference limens than the normal subjects, especially in frequency regions of greatest sensitivity loss. The masking noise produced a similar increase in frequency difference limens in both groups. Although several previous speech recognition studies have demonstrated an interactive effect between the presence of hearing loss and masking noise, the results of the present study demonstrate that frequency discrimination performance for normal-hearing and hearing-impaired subjects is not differentially affected by the addition of masking noise.La discrimination frequentielle a 500, 1 000 et 2 000 Hz a ete mesuree dans le silence et ...

Principle of spectral imagery by speckle and Fourier's interferometries

Abstract: Spectral imagery by speckle and Fourier's interferometries is a high spectral and spatial resolution technique. The authors simultaneously observe, in short exposure, at the focus of a telescope the two images, of a same astronomical source, which are transmitted and reflected by a Michelson interferometer in which the path difference moves step by step. They show that they can obtain the image of the source at all wavelengths of the studied spectral interval, with all spatial and spectral resolutions allowed respectively by the telescope and the Michelson interferometer. This method permits a spectral selectivity suitable to the studied object. They show the principle of this method in an ideal case, that when the detection noise is neglected. They study the effect of noise on the quality of the final result. The study of the effects of turbulence allows to specify the maximum allowed spectral band.

Receiver for sound multiplex broadcast

Abstract: A sound multiplex broadcast receiver wherein a SAP channel for a subchannel carrier wave audio signal is provided independently of a channel for the main audio signal to demodulate the subchannel audio signal by way of a noise reduction circuit. The receiver comprises a frequency characteristics correction circuit for correcting the frequency characteristics of the demodulated subchannel audio signal by diminishing the gain at a specific frequency where the distortion factor of the noise reduction circuit reaches a peak, and a noise detection circuit for detecting the level of noise in the audio signal received. The amount of correction by the correction circuit is controlled by the output of the noise detection circuit which varies with the intensity of electric field received, permitting the noise reduction circuit to produce a SAP signal output with flat frequency characteristics.

Method of timely controlling opening and closing times by means of a computer-controlled time code

Abstract: In stereo mixing, not all instruments which play on a piece of music are constantly active. The missing audio information on a channel has the effect that only the noise of the sound source and of the amplifiers go into the composite audio signal, thereby worsening the signal-to-noise ratio of the stereo mix. In order to avoid losses in quality, the sound engineer uses a switch to block manually all the channels currently not producing an audio signal. This manually controlled method is extremely time-consuming and requires great skill on the part of the sound engineer, the natural reaction time of the operator preventing precise control of the channels. The solution to the problem lies in a computer-controlled opening and closing of the music mixing desk channels. The first time the piece of music is played, the computer registers from the time code on the tape those times at which an audio signal occurs or is missing on a channel. By including the reaction times existing in the system, whenever the piece of music is played after that the computer can automatically and precisely carry out the controlling of the opening and closing operations by advancing the opening times or delaying the closing times by the amount of the reaction time. The main application of the invention is in the field of professional and semi-professional sound studio engineering. It leads on the one hand to a considerable reduction in the time spent in mixing and additionally provides an optimisation of the signal-to- noise ratio in the final stereo mix.

Relations between speech intelligibility and psychoacoustical modulation transfer function (PMTF).

Abstract: The modulation transfer function, MTF, has proved to be a powerful measure for predicting speech intelligibility in speech transmission channels. We extended it to include the ear, by measuring the psycho-acoustical MTF, i.e. the PMTF. Tone thresholds of 11 normal-hearing and 20 hearing-impaired subjects were measured in presence of unmodulated and intensity modulated noise. Octave frequencies from 500 to 4,000 Hz were used. The noise was octave filtered around the frequency of the probe tone. Six modulation frequencies from 1 to 50 Hz were used. From these results the PMTFs were calculated, as well as the corresponding psycho-acoustical speech transmission indices, i.e. the PSTIs. The subjects' speech discrimination scores in quiet and speech reception thresholds in noise were also measured. A correlation coefficient of 0.85 between the speech discrimination score in quiet and the PSTI was obtained. For the speech reception threshold in noise and the PSTI, the correlation was 0.71. The first of these two figures is promising, but our method needs some improvement, as it gave some problems due to fatigue effects.

Noise suppressing circuit for privacy communication equipment

Abstract: PURPOSE:To prevent the unpleasant feeling and the hearing sense obstacle of a user by holding down the sound volume of reproduced sounds if the audio signal subjected to the scrambling processing in a digital communication equipment is reproduced without the descrambling processing or the transmission side and the reception side are different in privacy code. CONSTITUTION:An audio input 16 from a transmitter or the like passes an amplifier 5 and is encoded by an AD converter 4 and is processed in a scrambling circuit 3 on a privacy code set preliminarily by a switch circuit 14 and is multiplexed with a control signal from a control signal transmitting and receiving circuit 13 and is transmitted to a digital line 15 through a transmission amplifier 1. The reception signal passes a reception amplifier 6, and the control signal and the audio signal are inputted to the transmitting and receiving circuit 13 and a descrambling circuit 8 respectively by a separating circuit 7, and the audio signal is processed with the privacy code set by the circuit 14 and is outputted to a receiver or the like through a DA converter 9, a sound volume suppressing circuit 10, and an amplifier 11. If it is judged that the audio signal after DA conversion is reproduced as a noise sound because of the difference of privacy code or the like, a detecting circuit 12 holds down the sound volume by the sound volume suppressing circuit 10.

Audio circuit system

Abstract: PURPOSE:To minimize the noise for no-speaking while keeping the transmission quality by detecting whether an audio input signal is present or not and transmitting forcibly a digital pattern corresponding to the no-speaking signal by a switch means if the input signal is absent. CONSTITUTION:When an analog audio signal is inputted, it is amplified by an amplifier 1, and the output is connected to an AD converter 2 and an audio detector 3. Though the audio signal converted to a digital signal is transmitted to an output through a digital switch 5, the digital switch 5 is switched to transmit the output of a pattern generator 4 is the input audio signal is absent. The digital switch 5 is switched by the control signal from the audio detector 3 so that the output of the AD converter 2 is transmitted for speaking and the digital signal corresponding to the no-speaking signal is transmitted from the pattern generator 4 for no-speaking.