Showing papers on "Noise published in 1993"

Uncomodulated glimpsing in ‘‘checkerboard’’ noise

Abstract: The ability of listeners to ‘‘glimpse’’ acoustic cues during the quieter sections of an interrupted noise has primarily been studied using maskers with interruptions occurring simultaneously across the entire frequency range of the masker—broadband comodulated interruptions. Here, the possibility of uncomodulated glimpsing (the glimpsing of acoustic cues separated both in time and frequency) was investigated. To achieve this, speech reception thresholds for a set of intervocalic consonants were adaptively measured in 100‐Hz to 10‐kHz pink noise divided into a varying number of frequency bands of equal energy. In uncomodulated noise conditions, the odd and even numbered bands were switched on and off alternately at a rate of 10 Hz. The spectrograms of such noises (on log frequency scales), resemble portions of a checkerboard. Glimpsing in ‘‘checkerboard’’ noise was found with maskers divided into two and four bands, but not into eight bands or more. Further investigations showed that, in the two‐band case, this release from masking was indeed due to uncomodulated glimpsing, and not simply attributable to glimpsing in one of the modulated bands. In the four‐band case, the release from masking in checkerboard noise can be accounted for without recourse to uncomodulated glimpsing. Interestingly, conditions which allowed glimpsing resulted in greater intersubject variability. The implications of these results for quantitative analyses of masker fluctuations are discussed.

Adaptive noise canceller

Abstract: An adaptive filter estimates the noise component in a primary audio signal by convolving a secondary audio signal with a set of coefficients. The estimated noise component is subtracted from the primary audio signal to produce an output signal. During steady-state operation, the adaptive filter coefficients are updated as to minimize the difference between the primary audio signal and the estimated noise component. Steady-state operation is identified automatically by monitoring the power level of the primary or secondary audio signal, or the power level of the output signal. Coefficient updating is suspended when the monitored power level rises from a steady state to an unsteady state, and is resumed when the monitored power level returns to its previous steady-state level, or settles into a new steady state.

Automatic volume controlling apparatus

Abstract: An apparatus for automatically controlling the sound volume of a sound producing apparatus based on ambient noise includes a microphone which detects the sound produced by the sound producing apparatus and ambient noise. Adaptive filters approximate, based on signals sent from the sound producing apparatus to speakers thereof, the characteristics of the sound components, relating to the sound produced by an individual speaker of the sound producing apparatus, which are received by the microphone. A subtractor subtracts the output of the adaptive filters from the output of the microphone to obtain a signal representing the ambient noise. The adaptive filters receive the signal representing the ambient noise and use this signal as a coefficient updating signal. A signal converter converts the signal representing ambient noise into an ambient noise level representing the volume of ambient noise. A controller then controls the gain by which amplifiers amplify signals sent from the sound producing apparatus to speakers thereof based on the ambient noise level.

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.

Frequency domain noise suppression approaches in mobile telephone systems

Abstract: Frequency-domain noise suppression systems with a single microphone are studied for adverse mobile noise environments. A novel noise suppression algorithm based on a modified maximum likelihood estimate is developed. The algorithm takes into account not only minimum voice distortion but also subjective criteria for the noise naturalness. A speech detector based on the temporal signal-to-noise ratio (SNR) information of every subband of a bandpass filter bank is proposed. The speech detector is proven to be very effective and robust in a rapidly changing noisy background. The proposed noise suppression method in mobile telephone systems demonstrates much better subjective and objective test results than the existing methods. A real-time test shows that 10 dB noise reduction and 9 dB SNR improvements have been achieved even in the most adverse mobile situation. >

Road traffic noise around schools: a risk for pupil's performance?

Abstract: Noise levels around educational centres can negatively affect the performance of both teachers and pupils. Two public schools in Valencia, Spain, were selected for study. One of these schools was exposed to excessively high road traffic noise levels while the other was located in a relatively quiet area. The socioeconomic level of those attending the schools was very similar. A set of external and internal noise measurements were carried out, along with two different attention tests among the children. Test results were consistently better (both for tests and for children from different classrooms in each school) in the quiet school. Exposure to high traffic noise levels in the noisy school over the whole school year is a plausible determinant of these results.

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.

Some acoustic effects of listening to noise on speech production

Abstract: An experiment was performed to corroborate earlier findings of a spectral shift in speech produced in 80 dB or greater noise, and to determine whether a spectral shift relative to quiet also occurs at 35‐ and 60‐dB white noise. Both speakers studied incrementally decreased spectral tilt (the amount of energy at low frequencies relative to high) with noise above 35 dB. Both speakers increased duration and average speech amplitude incrementally with each noise increase. One speaker steadily increased fundamental frequency, while the other speaker decreased it. Speech changes are discussed as possible compensations for a distorted sidetone.

Auditory spectral discrimination and the localization of clicks in the sagittal plane.

Abstract: Experiments show that the ability of human listeners to localize an impulsive sound in the medial sagittal plane (front, overhead, rear) deteriorates as the level of the sound increases. This negative level effect is strong for clicks but does not appear for broadband noise. It is conjectured that the negative level effect arises because the tonotopic excitation pattern is broadened for intense impulsive sounds. As a result, the spectral peaks and valleys, which are caused by anatomical filtering and which normally code for localization in the sagittal plane, are less recognizable. Filtered click discrimination experiments using headphones also show a negative level effect for clicks, but not for noise, and support this conjecture.

Adaptive proportional gain audio mixing system

Abstract: An improved audio mixing system. The system is not limited to one type of microphone and does not depend on separate sensing microphones to determine the background noise level. In addition, the active signal is not amplitude modulated by extraneous noise inputted through inactive microphones. The mixing system divides the system gain between a plurality of input channels. The mixing system periodically samples the channels and determines which channel is the dominant channel. In response to this determination, the gain of the dominant channel is increased and that of the other channels decreased.

High efficiency encoding device and a noise spectrum modifying device and method

Abstract: An apparatus and method for processing an input digital signal in which an input digital audio signal at an input terminal 11 is supplied via a one-block delay unit 17 to a difference unit 20, an output of which is processed by a noise-shaping circuit 21 and restored by a near-instantaneous compressor 26, a near-instantaneous expander 27 and an accumulator 28 so as to be fed back to the difference unit 20 to find a difference between a signal of a directly preceding sample and a signal of the current sample. The noise shaping circuit 21 modifies the spectrum of the quantization noise by taking aural characteristics, such as equal-loudness characteristics or masking characteristics, into account for diminishing the level of the quantization noise as perceived by the ear. The noise level as perceived may be reduced in a manner desirable for the sound quality without changing the existing format.

Evaluating a speech‐reception threshold model for hearing‐impaired listeners

Abstract: Plomp's speech reception threshold (SRT) model [R Plomp, J Acoust Soc Am 63, 533-549 (1978)] incorporates a distortion and an attenuation factor that are both expressed in dB and, for hearing-impaired listeners, are greater than 0 dB The distortion factor is hypothesized to affect the SRT in quiet and in noise and suggests that a hearing-impaired listener will always demonstrate a higher SRT than a normal-hearing listener The present study examines whether this distortion factor can be explained for many listeners simply by inaudibility of a portion of the speech spectrum SRTs were obtained from normal-hearing and mild-to-moderately hearing-impaired listeners in quiet and at various noise levels The results indicate that, at high noise levels, when the noise, rather than the quiet threshold, becomes the dominating factor, the SRT functions of both the normal and the mild-to-moderately hearing-impaired group converge and the distortion factor diminishes to zero Predictions were also made using an articulation index and similar convergence of the two functions was observed

Stethoscopic system for high-noise environments

Abstract: An electronic stethoscopic system is described which permits detection of auscultory sounds in a patient in high noise environments such as ambulances and aircraft. The stethoscope employs an electroacoustical transducer, an acoustical driver mounted in a headset providing acoustical isolation from exterior noise, a summing microphone positioned within the insulating headset, and active noise reduction circuitry to feed an error signal back from the summing microphone to the acoustical driver so as to effectively cancel the unwanted acoustical noise originating external to the insulating headset. The stethoscopic system includes circuitry permitting the headset to selectively receive the audio output from a vehicular intercom system whenever a voice signal is present, thereby allowing treating medical personnel to monitor the patient while participating in the conversation being conducted on the vehicle's intercom system.

Non-intrusive speech level and dynamic noise measurements

Abstract: A digital signal processor (115) is provided with the facility to measure accurately, and nonintrusively, the levels of noise and/or speech signals appearing on an in-service network connection (310).

Noise suppression in digital voice communications systems

Abstract: Noise is suppressed during pauses and silent periods in conversation, and voiced signals are freely passed in a mobile communications system thereby improving the quality of the transmitted speech. Noise suppression is implemented in such a way that a low level of noise is still allowed to pass to provide presence of the remote speaker; that is, the line is not made completely silent as this may falsely indicate that the connection has been interrupted. The noise suppression has an added feature of decreasing the background noise fractionally when voice is no longer detected. This provides perceptually improved quality of the communication.

The recognition and interpretation of idioms 1

Abstract: 1 John turned on the light 2 Joe put up with the noise 3 Bill took advantage of her generosity 4 He did it for his mother’s sake 5 The old curmudgeon finally kicked the bucket 6 We got him to spill the beans 7 That really put the cat among the pigeons 8 He was chasing a red herring 9 We feel they are skating on thin ice with that project 10 He shouldn’t have counted his chickens before they were hatched

Method and apparatus for editing an audio signal

Abstract: An audio signal editing method and apparatus for use with original sound data which is already compressed and encoded in predictive form, for example ADPCM. To edit a desired portion from the original audio data, the method and apparatus reconstruct (encoded PCM) audio data from the original (ADPCM) audio data and encode a non-predictive unit to replace a first original audio data unit of the desired portion. Thus, the desired portion is rendered independent of preceding sample values, without having to be recoded entirely. An enhanced method and apparatus encode further replacement units to eliminate noise introduced by the basic method.

Suppression of impulsive disturbances from audio signals

Abstract: A simple and effective method is proposed for suppressing impulsive noise from audio signals with minimal high frequency loss. Suppression is accomplished by median filtering the contaminated signal sections. A gating signal that enables local median filtering is generated by an algorithm that involves impulse detection and recursive median filtering.

Sleep induction effects of steady 60 dB (A) pink noise.

Abstract: The effects of 60 dB (A) steady pink noise on sleep induction were studied Two experiments were conducted First, 10 night-sleep polygrams of a young male subject were recorded consecutively as controls Five night polygrams of the same subject were then recorded with exposure to steady 60 dB (A) pink noise Second, polygrams of four students were recorded using the same noise exposure as in the first experiment Polygrams for the control night were also recorded Noise exposure tended to shorten sleep latency, ie, the values were 42 and 95 min in the first and second experiment, respectively The steady 60 dB (A) noise made sleep induction easier

The influence of nocturnal aircraft noise on sleep and on catecholamine secretion.

Abstract: The influence of noise from night flying on electro-biological reactions and on the secretion of catecholamines (adrenaline and noradrenaline) was studied in eight healthy adults whose place of residence exposes them to day-time aircraft noise. The interrelationships were then analysed, with daytime noise exposure, personality traits and general day-to-day condition reflected in control variables. The subjects were each observed during five nights without noise exposure (Leq Language: de

Method of compressing an audio signal using adaptive bit allocation taking account of temporal masking

Abstract: An audio signal is coded by converting the audio signal into a signal in a frequency domain and effecting a big allocation on the converted audio signal. A masking threshold level for reducing aurally recognized noise due to a masking effect is determined using a signal spectral distribution in a present frame and a signal spectral distribution in a past frame. If the difference between the determined masking threshold level and a masking threshold level in the past frame is equal to or greater than a predetermined level, then a level limited to the difference corresponding to the predetermined level is regarded as a masking threshold level in the present frame.

Short wavelength mo recording: system noise issues and performance optimization

Abstract: Optical recording at 488 nm using optimized CoPt multilayer media is described. Large mark CNR values of 62 dB are achieved by ensuring that system noise is dominated by the media noise component through a combination of media desensitization, read power optimization., and, elimination of the groove noise component which usually dominates the system noise. The results indicate that it is generally not possible to decrease the laser power requirements when utilizing a small recording laser spot size despite the increased power density because of these noise constraints. Further, because of a decreased detector sensitivity at blue recording wavelengths, further desensitization beyond that necessary to compensate for the increase power density is required when shorter wavelengths are used to achieve a smaller spot size.

Car audio system with high signal output

Abstract: A circuit for coupling a 2- or 4-channel audio deck to 2- or 4-speakers in a car stereo audio system to increase the signal level output without raising the supply voltage or increasing the noise The circuit comprises unity gain inverting and non-inverting amplifiers each connected in one of four channels, with the outputs combined to provide left and right channel outputs for series connection, respectively, to the left and right channel speakers

Comparison of Noniterative Factor Analysis Spectral Resolution Algorithms

Abstract: The effects of fundamental and ancillary algorithm differences on the performance of three noniterative factor analysis spectral resolution algorithms on noisy and overlapped bilinear matrix-formatted spectral data are evaluated and compared. The evaluation consists of the analysis of simulated fluorescence excitation-emission matrices in which the spectral overlap, noise type, and level were systematically varied. The results indicate that the conventions used to exclude low-intensity, high-noise rows and columns from consideration as component spectra estimates and to choose the first estimates of the component spectra have significant impact on resolution algorithm performance. The results of the application of the algorithms to ideal data are nearly identical; however, there are several distinctions in the performance of the algorithms on noisy data. Verifiable estimates of the component spectra were resolved from data matrices degraded by white and Poisson noise that have signal-to-noise (S/N) ratios above 10 by all three algorithms regardless of the noise level and the degree of spectral overlap. The impact of pink noise was uniformly deleterious at S/N below 15.

Audio peripheral mixer circuit and method for noise reduction

Abstract: A circuit and method for mixing signals from audio sources provide noise reduction by employing a noise gate to selectively attenuate one of the source signals. The noise gate has a low gain state and a unity gain state. The noise gate switches to its unity gain state when a control signal reaches a predefined threshold voltage. The control signal is generated by rectifying and integrating the noisy source audio signal, so that the noise gate enters the unity gain state only when a useful sound occurs. Thus, the noise in the combined audio signal is reduced, without attenuating useful components of the source signals.

A study on recording noise with readback waveform analysis

Abstract: The recording noise of sputtered Co alloy media at high recording density was studied with readback waveform. High noise power of longitudinal recording media at high recording density was correlated to the fluctuation of magnetic transition region. In perpendicular recording media, in contrast, the fluctuation of magnetic transition region was small even at high recording density, which leads to low noise power.

Analysis of subband quantization noise level and shapes: a function of wideband audio codec tandemming

Abstract: Newly developed codecs for wideband audio signals rely on subband coding and psychoacoustic modeling to achieve significant reduction in the bit rate of a digital audio signal. An investigation into the shape and level of quantization noise introduced into a coded audio signal is performed. Psychoacoustic principles reveal that quantization noise is masked, i.e. made inaudible, if its spectral shape fails below that of the original signal by a 13 dB threshold. Simulation results show that there is a strong correlation of quantization noise and original signal shapes at low bit allocations. Quantization noise levels which increase result in a degradation of the quality of a codec. A criterion for the evaluation of a codec is its performance when tandemmed. Simulation results reveal that there can be up to an 18 dB increase in average subband noise energy for eight tandems. Surprisingly, subbands more vulnerable to an increase in quantization noise tend to exhibit a high signal to mask ratio, as determined by the psychoacoustic model. >

The effect of amplitude comodulation on extracting sentences from noise: Evidence from a variety of contexts

Abstract: While signal‐to‐noise ratios range from about 50–90 dB in laboratory experiments, measurements of speech levels in most natural environments show much poorer listening conditions. For example, Teder [Hear. Instrum. 11, 32–33 (1990)] measured signal‐to‐noise ratios ranged from a high of 13 dB in a carpeted office to low of 1 dB in a 1986 Chevy Nova traveling at 55 mph. These numbers indicate an immense difference between speech signals presented to listeners in laboratory experiments as opposed to the real‐world environment. Although speech intelligibility has been well studied in noise, there has been little study of the acoustic characteristics of speech that allow the message to be separated from the background noise. One characteristic of voiced speech that shows great promise in this regard is amplitude comodulation. In the present research, amplitude comodulated tone‐analog sentences [Carrell and Opie, Percept. Psychophys. 52, 437–445 (1989)] were presented with simultaneous white noise and multispeaker babble at 0‐, 5‐, and 10‐dB signal‐to‐noise ratios. It was found that the beneficial effect of amplitude comodulation was greater at the lower signal‐to‐noise ratios.

Integrated Hands-free Cellular Phone, Active Noise Control And Audio System

Abstract: This paper presents an integrated system which provides adaptive noise cancellation and acoustic echo cancellation capabiities for hands-free cellular phones and also perfoms active noise control to reduce the noise inside an automobile. The Musical Interference Suppression (MIS) filter is used to reduce the interference of the music in updating the coefficients of the adaptive filters. Further, this integrated audio system reduces the overall cost and provides better on-line e m path and echo path modeling. Introduction Cellular (mobile) phone has been widely used in recent years. The hands free operation offers users a safer and more convenient way of using cellular phones while driving a car. Active noise control (ANC) is based on the principle of superposition i.e., an anti-sound of equal in amplitude and opposite in phase is generated and acoustically combined with an unwanted sound, thus resulting in the cancellation of both the sounds. Three dimensional active noise control systems have many applications such as, in the interior of the trucks, cars and localized quiet zones. However, the hands-free system that uses microphone and loudspeaker instead of handset has two major difficulties: high volume of noise and acoustic echo interferences. Therefore, two adaptive filters are required to cancel the line (electrical) echo and the acoustic echo and the third filter is needed to cancel the ambient noise. In the proposed integrated system, the ANC system reduces the engine related noise inside the passenger compartment. This integrated system uses the existing audio components such as amplifiers, speakers, power amplifiers for the 3-D active noise control thus minimizing the redundancies of the system. The MIS filter estimates the true error signal that is necessary for the hands-free phone and the ANC system. Therefore the problems of misadjustment from the interfekg audio signals to the system is therefore solved by the MIS process. Furthermore, the MIS process estimates the error and echo paths on-line which can improve the performance of the adaptive echo cancellers, noise canceller, and the ANC system. 270 Integrated System The pposed system integrates the hauds-free cellular phone with the existing car audio system. It has active noise control capabiity which generates a quiet zone that surrounds the driver's head rest. This integrated system operates in either one of the following two modes: 1. Active noise control mode the system performs threedimensional active noise control to reduce engine-related noise inside the car. 2. Hands-free phone (HFP) mode the system performs both line echo and acoustic echo cancellation to provide full-duplex capability. The system also performs adaptive noise cancellation to reduce the background noise picked up by the microphone. When the cellular phone is used, the active noise control system is disabled, therefore, a single DSP chip can provide the necessary computational requirement. As illustrated in Figure 1, the integrated system is connected to a regular cellular phone and car audio system. By the integration of existing car audio system, only two extra microphones are needed for the voice pickup in HFP mode, or residual noise pickup to update the active noise control filters in ANC mode. Since the power amplifier and loud speakers of the audio system can be used, the cost of the overall system is greatly reduced. Furthermore, the integrated system can suppress the interference of music in both the HFP and ANC modes. Figure 1. Integrated Hands-Free Cellular Phone, Active Noise Control and Audio System 0-7803-0843-3/93 $3.00