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Showing papers on "Background noise published in 1996"


Journal ArticleDOI
TL;DR: In this paper, the authors present a technique for isolating climate signals in time series with a characteristic "red" noise background which arises from temporal persistence, which is estimated by a robust procedure that is largely unbiased by the presence of signals immersed in the noise.
Abstract: We present a new technique for isolating climate signals in time series with a characteristic ‘red’ noise background which arises from temporal persistence. This background is estimated by a ‘robust’ procedure that, unlike conventional techniques, is largely unbiased by the presence of signals immersed in the noise. Making use of multiple-taper spectral analysis methods, the technique further provides for a distinction between purely harmonic (periodic) signals, and broader-band (‘quasiperiodic’) signals. The effectiveness of our signal detection procedure is demonstrated with synthetic examples that simulate a variety of possible periodic and quasiperiodic signals immersed in red noise. We apply our methodology to historical climate and paleoclimate time series examples. Analysis of a ≈ 3 million year sediment core reveals significant periodic components at known astronomical forcing periodicities and a significant quasiperiodic 100 year peak. Analysis of a roughly 1500 year tree-ring reconstruction of Scandinavian summer temperatures suggests significant quasiperiodic signals on a near-century timescale, an interdecadal 16–18 year timescale, within the interannual El Nino/Southern Oscillation (ENSO) band, and on a quasibiennial timescale. Analysis of the 144 year record of Great Salt Lake monthly volume change reveals a significant broad band of significant interdecadal variability, ENSO-timescale peaks, an annual cycle and its harmonics. Focusing in detail on the historical estimated global-average surface temperature record, we find a highly significant secular trend relative to the estimated red noise background, and weakly significant quasiperiodic signals within the ENSO band. Decadal and quasibiennial signals are marginally significant in this series.

1,143 citations


Journal ArticleDOI
14 Mar 1996-Nature
TL;DR: It is demonstrated that broadband stochastic resonance is manifest in the peripheral layers of neural processing in a simple sensory system, and that it plays a role over a wide range of biologically relevant stimulus parameters.
Abstract: SENSORY systems are often required to detect a small amplitude signal embedded in broadband background noise. Traditionally, ambient noise is regarded as detrimental to encoding accuracy. Recently, however, a phenomenon known as stochastic resonance has been described in which, for systems with a nonlinear threshold, increasing the input noise level can actually improve the output signal-to-noise ratio over a limited range of signal and noise strengths. Previous theoretical and experimental studies of stochastic resonance in physical1–7and biological6–10 systems have dealt exclusively with single-frequency sine stimuli embedded in a broadband noise background. In the past year it has been shown in a theoretical and modelling study that stochastic resonance can be observed with broadband signals11,12. Here we demonstrate that broadband stochastic resonance is manifest in the peripheral layers of neural processing in a simple sensory system, and that it plays a role over a wide range of biologically relevant stimulus parameters. Further, we quantify the functional significance of the phenomenon within the context of signal processing, using information theory.

667 citations


Journal ArticleDOI
TL;DR: In this paper, an Occam's inversion algorithm for crosshole resistivity data that uses a finite-element method forward solution is discussed, where the earth is discretized into a series of parameter blocks, each containing one or more elements.
Abstract: An Occam's inversion algorithm for crosshole resistivity data that uses a finite-element method forward solution is discussed. For the inverse algorithm, the earth is discretized into a series of parameter blocks, each containing one or more elements. The Occam's inversion finds the smoothest 2-D model for which the Chi-squared statistic equals an a priori value. Synthetic model data are used to show the effects of noise and noise estimates on the resulting 2-D resistivity images. Resolution of the images decreases with increasing noise. The reconstructions are underdetermined so that at low noise levels the images converge to an asymptotic image, not the true geoelectrical section. If the estimated standard deviation is too low, the algorithm cannot achieve an adequate data fit, the resulting image becomes rough, and irregular artifacts start to appear. When the estimated standard deviation is larger than the correct value, the resolution decreases substantially (the image is too smooth). The same effects are demonstrated for field data from a site near Livermore, California. However, when the correct noise values are known, the Occam's results are independent of the discretization used. A case history of monitoring at an enhanced oil recovery site is used to illustrate problems in comparing successive images over time from a site where the noise level changes. In this case, changes in image resolution can be misinterpreted as actual geoelectrical changes. One solution to this problem is to perform smoothest, but non-Occam's, inversion on later data sets using parameters found from the background data set.

459 citations


Journal ArticleDOI
TL;DR: Evidence suggests that a number of adverse effects of noise in general arise from exposure to low-frequency noise: Loudness judgments and annoyance reactions are sometimes reported to be greater for low- frequency noise than other noises for equal sound-pressure level.
Abstract: The sources of human exposure to low-frequency noise and its effects are reviewed. Low-frequency noise is common as background noise in urban environments, and as an emission from many artificial sources: road vehicles, aircraft, industrial machinery, artillery and mining explosions, and air movement machinery including wind turbines, compressors, and ventilation or air-conditioning units. The effects of low-frequency noise are of particular concern because of its pervasiveness due to numerous sources, efficient propagation, and reduced efficacy of many structures (dwellings, walls, and hearing protection) in attenuating low-frequency noise compared with other noise. Intense low-frequency noise appears to produce clear symptoms including respiratory impairment and aural pain. Although the effects of lower intensities of low-frequency noise are difficult to establish for methodological reasons, evidence suggests that a number of adverse effects of noise in general arise from exposure to low-frequency noise: Loudness judgments and annoyance reactions are sometimes reported to be greater for low-frequency noise than other noises for equal sound-pressure level; annoyance is exacerbated by rattle or vibration induced by low-frequency noise; speech intelligibility may be reduced more by low-frequency noise than other noises except those in the frequency range of speech itself, because of the upward spread of masking. On the other hand, it is also possible that low-frequency noise provides some protection against the effects of simultaneous higher frequency noise on hearing. Research needs and policy decisions, based on what is currently known, are considered.

410 citations


Journal ArticleDOI
TL;DR: It is suggested that recent studies based on a Source Generator Framework can provide a viable foundation in which to establish robust speech recognition techniques, and three novel approaches for signal enhancement and stress equalization are considered to address the issue of recognition under noisy stressful conditions.

270 citations


Patent
Peter Händel1
12 Jan 1996
TL;DR: In this paper, a spectral subtraction noise suppression method in a frame-based digital communication system is described, where each frame includes a predetermined number N of audio samples, thereby giving each frame N degrees of freedom.
Abstract: A spectral subtraction noise suppression method in a frame based digital communication system is described. Each frame includes a predetermined number N of audio samples, thereby giving each frame N degrees of freedom. The method is performed by a spectral subtraction (150) function H(φ) which is based on an estimate (140) Ζv(φ) of the power spectral density of background noise of non-speech frames and an estimate (130) Ζx(φ) of the power spectral density of speech frames. Each speech frame is approximated (120) by a parametric model that reduces the number of degrees of freedom to less than N. The estimate Ζx(φ) of the power spectral density of each speech frame is estimated (130) from the approximative parametric model.

218 citations


Journal ArticleDOI
TL;DR: This method provides consistent noise estimates from images with very different land cover types and works well with inhomogeneous images (e.g., of a vegetated area such as Jasper Ridge) unlike a method described recently by Gao.
Abstract: A new method is presented for computing the noise affecting each band of an AVIRIS hyperspectral image. Between-band (spectral) and within-band (spatial) correlations are used to decorrelate the image data via linear regression. Each band of the image is divided into small blocks, each of which is independently decorrelated. The decorrelation leaves noise-like residuals whose variance estimates the noise. A homogeneous set of these variances is selected and their values are combined to provide the best estimate of that band's noise. This method provides consistent noise estimates from images with very different land cover types. Its performance is validated by comparing its noise estimates with noise measures provided with two AVIRIS images. The method works well with inhomogeneous images (e.g., of a vegetated area such as Jasper Ridge) unlike a method described recently by Gao. The method is automatic and does not require the intervention of a human operator. Noise estimates are presented for 10...

199 citations


Journal ArticleDOI
TL;DR: Segment analysis, using magnitude-squared coherence (MSC) or related statistics, has equivalent statistical power; MSC and F each yield unbiased SNR estimates that have identical distributions when SNR = 0.
Abstract: Sinusoids in background noise can conveniently be detected using unsegmented power spectra, comparing power at the signal frequency to average power at several neighbor frequencies. In this case, the F test is preferable to t tests based on rms or dB values, because of the skewed distributions of rms and dB when signal‐to‐noise ratio (SNR)=0. F‐test performance improves as the number of frequencies increases, to about 15, but can be degraded if the background noise is not white, with a slope exceeding about 10 dB for the range of frequencies sampled. Segment analysis, using magnitude‐squared coherence (MSC) or related statistics, has equivalent statistical power; MSC and F each yield unbiased SNR estimates that have identical distributions when SNR=0. Selection of F or MSC for detection of sinusoids will usually be a matter of convenience.

169 citations


Proceedings ArticleDOI
07 May 1996
TL;DR: The enhanced MELP speech coder is described, which is a candidate for the new U.S. Federal Standard at 2.4 kbits/s and has been optimized for performance in acoustic background noise and in channel errors, as well as for efficient real-time implementation.
Abstract: This paper describes our enhanced mixed excitation linear prediction (MELP) speech coder which is a candidate for the new U.S. Federal Standard at 2.4 kbits/s. The new coder is based on the MELP model, and it uses a number of enhancements as well as efficient quantization algorithms to improve performance while maintaining a low bit rate. In addition, the coder has been optimized for performance in acoustic background noise and in channel errors, as well as for efficient real-time implementation. Listening tests confirm that the enhanced 2.4 kbit/s MELP coder performs as well as the higher bit rate 4.8 kbit/s FS1016 CELP standard.

169 citations


Journal ArticleDOI
TL;DR: Both models possess resonant phenomena in the first passage probability distribution and mean first passage time, arising from the interplay of characteristic time scales in the system.
Abstract: The dynamics of the standard integrate-fire model and a simpler model (that reproduces the important features of the integrate-fire model under certain conditions) of neural dynamics are studied in the presence of a deterministic external driving force, taken to be time-periodic, and white background noise. Both models possess resonant phenomena in the first passage probability distribution and mean first passage time, arising from the interplay of characteristic time scales in the system. \textcopyright{} 1996 The American Physical Society.

159 citations


Journal ArticleDOI
TL;DR: In this article, the authors used a deep (1500 m) cased borehole near the town of Datil in west-central New Mexico to study high-frequency (>1 Hz) seismic noise characteristics.
Abstract: We used a deep (1500 m) cased borehole near the town of Datil in west-central New Mexico to study high-frequency (>1 Hz) seismic noise characteristics. The remote site had very low levels of cultural noise, but strong winds (winter and spring) made the site an excellent candidate to study the effects of wind noise on seismograms. Along with a three-component set of surface sensors (Teledyne Geotech GS-13), a vertical borehole seismometer (GS-28) was deployed at a variety of depths (5, 43, and 85 m) to investigate signal and noise variations. Wind speed was measured with an anemometer. Event-triggered and time-triggered data streams were recorded on a RefTek 72-02 data acquisition system located at the site. Our data show little cultural noise and a strong correlation between wind speed and seismic background noise. The minimum wind speed at which the seismic background noise appears to be influenced varies with depth: 3 m/sec at the surface, 3.5 m/sec at 43 m in depth, and 4 m/sec at 85 m in depth. For wind speed below 3 to 4 m/sec, we observe omni-directional background noise that is coherent at frequencies below 15 Hz. This coherence is destroyed when wind speeds exceed 3 to 4 m/sec. We use a test event ( Md ∼ 1.6) and superimposed noise to investigate signal-to-noise ratio (SNR) improvement with sensor depth. For the low Q valley fill of the Datil borehole (DBH) site, we have found that SNR can be improved by as much as 20 to 40 dB between 23 and 55 Hz and 10 to 20 dB between 10 and 20 Hz, by deploying at a 43-m depth rather than at the surface. At the surface, there is little signal above noise in the 23- to 55-Hz frequency band for wind speeds greater than 8 m/sec. Thus, high-frequency signal information that is lost at the surface can be recorded by deploying at the relatively shallow depth of 40 m. Because we observe only minor further reductions in seismic background noise (SBN) at deeper depths, 40 m is likely to be a reasonable deployment depth for other high-frequency-monitoring sites in similar environmental and geologic conditions.

Journal ArticleDOI
TL;DR: In this paper, a discrimination matching experiment was conducted to find the point where their perceived tone/noise ratios are the same, and it was shown that the matching standard increases 3.8 dB per doubling of the number of iterations in the IRN stimulus.
Abstract: Rippled noise is constructed by delaying a random noise and adding it back to the original. Iterated rippled noise (IRN) is constructed by repeating the delay‐and‐add process. IRN produces a two‐component perception, i.e., a buzzy tone with a pitch equal to the reciprocal of the delay and a background noise that sounds like the original random noise. The perceived tone/noise ratio increases with the number of iterations. The effective tone/noise ratio in IRN sounds with 1–16 iterations was measured in a discrimination matching experiment; each IRN was paired with a range of standard sounds, having varying proportions of a broadband noise and a complex tone, to find the point where their perceived tone/noise ratios are the same. The experiment shows that the tone/noise ratio of the matching standard increases 3.8 dB per doubling of the number of iterations in the IRN stimulus. Spectral models of auditory perception explain the pitch of IRN in terms of peaks in the region of the first five to eight harmonic...

Journal ArticleDOI
TL;DR: Significant advances have been made during the past 45 years in the understanding of airplane interior noise, and methods for the prediction of interior noise are summarized in this article, including recent work in active noise control.

Patent
08 Feb 1996
TL;DR: The spatial sound conference system as discussed by the authors enables participants in a teleconference to distinguish between speakers even during periods of interruption and overtalk, identify speakers based on spatial location cues, understand low volume speech, and block out background noise using spatial sound information.
Abstract: The spatial sound conference system enables participants in a teleconference to distinguish between speakers even during periods of interruption and overtalk, identify speakers based on spatial location cues, understand low volume speech, and block out background noise using spatial sound information. Spatial sound information may be captured using microphones positioned at the ear locations of a dummy head at a conference table, or spatial sound information may be added to a participant's monaural audio signal using head-related transfer functions. Head-related transfer functions simulate the frequency response of audio signals across the head from one ear to the other ear to create a spatial location for a sound. Spatial sound is transmitted across a communication channel, such as ISDN, and reproduced using spatially disposed loudspeakers positioned at the ears of a participant. By inserting a spatial sound component in a teleconference, a speaker other than the loudest speaker may be heard during periods of interruption and overtalk. Additionally, speakers may be more readily identified when they have a spatial sound position, and the perception of background noise is reduced.

Patent
05 Dec 1996
TL;DR: In this article, a method of noise suppression, a mobile station and a noise suppressor for suppressing noise in a speech signal was proposed, which consisted of means (20, 50) for dividing the speech signal into a first amount of subsignals and suppression means (30) for suppressing a subsignal (X, P) based upon a determined suppression coefficient (G).
Abstract: The invention relates to a method of noise suppression, a mobile station and a noise suppressor for suppressing noise in a speech signal. The suppressor comprises means (20, 50) for dividing the speech signal into a first amount of subsignals (X, P), which subsignals represent certain first frequency ranges, and suppression means (30) for suppressing noise in a subsignal (X, P) based upon a determined suppression coefficient (G). The noise suppressor further comprises recombination means (60) for recombining a second amount of subsignals (X, P) into a calculation signal (S), which represents a certain second frequency range, which is wider than the first frequency ranges and determination means (200) for determining a suppression coefficient (G) for the calculation signal (S) based upon the noise contained by it. The suppression means (30) are arranged to suppress the subsignals (X, P) recombined into the calculation signal (S) by said suppression coefficient (G), determined based upon the calculation signal (S).

Proceedings ArticleDOI
07 May 1996
TL;DR: Different methods of combining the visual and acoustic data to improve the recognition performance of automated speech recognizers by using additional visual information are presented, achieving error reduction of up to 50%.
Abstract: We present work on improving the performance of automated speech recognizers by using additional visual information: (lip-/speechreading); achieving error reduction of up to 50%. This paper focuses on different methods of combining the visual and acoustic data to improve the recognition performance. We show this on an extension of an existing state-of-the-art speech recognition system, a modular MS-TDNN. We have developed adaptive combination methods at several levels of the recognition network. Additional information such as estimated signal-to-noise ratio (SNR) is used in some cases. The results of the different combination methods are shown for clean speech and data with artificial noise (white, music, motor). The new combination methods adapt automatically to varying noise conditions making hand-tuned parameters unnecessary.

Journal ArticleDOI
TL;DR: In this article, the authors presented unthresholded, integrated data of micronekton and Euphausia superba collected with a SIMRAD EK500 at frequencies of 38 and 120 kHz, where background noise level follows a 20 log R+2αR relationship, which can be scaled to the minimum volume backscatter (Sv) in each layer during a transect and then subtracted from the entire data set to remove background noise.
Abstract: Echo integration of biological organisms with a low target strength can be difficult because of the problem of setting suitable pre-integration thresholds, and this is particularly acute with higher frequencies, such as 120 or 200 kHz, which are often used in studies of euphausiids. One solution is to integrate data without any threshold and then remove background noise during post-processing. Unthresholded, integrated data of micronekton and Euphausia superba collected with a SIMRAD EK500 at frequencies of 38 and 120 kHz are presented. The underlying background noise level follows a 20 log R+2αR relationship, which can be scaled to the minimum volume backscatter (Sv) in each layer during a transect and then subtracted from the entire data set to remove background noise. The utility of this procedure is demonstrated by making comparisons of Sv at each frequency and investigating the effect of noise removal on the identification of targets based on the dB difference (120 kHz Sv–38 kHz Sv).

PatentDOI
TL;DR: An active noise control stethoscope enables a physician or paramedic to check vital signs in the presence of high background noise levels as mentioned in this paper. But it is not suitable for use in medical applications.
Abstract: An active noise control stethoscope enables a physician or paramedic to check vital signs in the presence of high background noise levels. A digital processing technique is used to remove noise from the output of a main detection sensor, the detector being impedance mismatched with air and therefore less sensitive to external airborne noise. Instead of a microphone, the detector uses a piezoceramic transflexural actuator mounted in a cylindrical piece of brass, with a polyurethane coating placed over the active side of the sensor to keep the sensor waterproof and broaden the response of the sensor. An identical sensor is placed above the device to detect background noise adjacent the device, the signals being combined to obtain a signal free of background noise. A third sensor is also used to electronically remove noise detected by the main sensor, the third sensor being positioned to pick-up noise coupled through the patient's body. The time varying voltages from signals output by these sensors are digitized and processed by the digital signal processor, and the output used to drive the speakers in the headset. The digital signal processor uses a least mean squared algorithm to digitally subtract out the part of the detector signal that is correlated to the signals from the second and third sensors. In addition, noise penetrating the earcups of the headset is reduced by using the speakers to generate antinoise. The antinoise is generated by a filtered X-adaptive digital algorithm, and also by a random noise cancellation system which sets up an infinite impulse response filter in which the coefficients are continually updated for minimizing an ear sensitivity weighted sound pressure level detected by a microphone inside the headset.

Patent
Karim Jamal1
08 Mar 1996
TL;DR: In this paper, a method and system for transmitting background noise information on a packet reservation multiple access radio channel was proposed, where relative priorities were assigned to background noise data transmissions and speech data, or other data traffic, transmissions.
Abstract: A method and system for transmitting background noise information on a packet reservation multiple access radio channel In the method and system, relative priorities are assigned to background noise data transmissions and speech data, or other data traffic, transmissions The priorities are assigned so that the effect of background noise data transmissions on the delays and quality of speech data, or other data traffic, transmissions within the system is reduced

Journal ArticleDOI
TL;DR: A fully automatic, hands‐off algorithm for calculation of a noise model is developed and replaces the previous manual and subjective weeding out of poor data which was based on the experience of the interpreter.
Abstract: Knowledge of the noise level and the nature of the noise is critical when processing transient electromagnetic (TEM) sounding data. An inadequate noise estimation may result in an erroneous interpr...

Journal ArticleDOI
TL;DR: An objective quantitative approach to the decision of when to stop averaging sweeps in auditory brain-stem response (ABR) testing is presented and it is quite possible to automate the procedure and the decision process.
Abstract: An objective quantitative approach to the decision of when to stop averaging sweeps in auditory brain‐stem response (ABR) testing is presented. This decision is based on (1) the knowledge of the amplitude distributions of wave V in the ABRs of normal hearing individuals for varying stimulus levels, (2) calculated estimates of the residual background noise in the average, and (3) use of a quantitative statistical detector of an evoked potential. Several reasons for terminating an average are presented along with a specific protocol for each of the reasons. These protocols provide a general but consistent framework to address the issue of when to stop averaging and should improve the efficiency of ABR testing. Furthermore, it is quite possible to automate the procedure and the decision process.

01 May 1996
TL;DR: The U.S. Department of Transportation, Research and Special Programs Administration, John A. Volpe National Transportation Systems Center, Acoustics Facility, in support of the Federal Highway Administration (FHWA), Office of Environment and Planning, has developed this document, which reflects significant improvements and changes in noise measurement technologies that have evolved since the 1981 FHWA publication, "Sound Procedures for Measuring Highway Noise". as discussed by the authors.
Abstract: The U.S. Department of Transportation, Research and Special Programs Administration, John A. Volpe National Transportation Systems Center, Acoustics Facility, in support of the Federal Highway Administration (FHWA), Office of Environment and Planning, has developed this document, which reflects significant improvements and changes in noise measurement technologies that have evolved since the 1981 FHWA publication, "Sound Procedures for Measuring Highway Noise". This report documents the recommended procedures for the measurement of (1) existing noise; (2) vehicle noise emissions; (3) barrier insertion loss; (4) construction equipment noise; (5) noise reduction due to buildings; and (6) occupational noise exposure.

Journal ArticleDOI
TL;DR: In this paper, the authors study the design of constant false-alarm rate (CFAR) tests for detecting a rank-one signal in the presence of background Gaussian noise with unknown spatial covariance.
Abstract: We study the design of constant false-alarm rate (CFAR) tests for detecting a rank-one signal in the presence of background Gaussian noise with unknown spatial covariance. We look at invariant tests, i.e., those tests whose performance is independent of the nuisance parameters, like the background noise covariance. Such tests are shown to have the desirable CFAR property. We characterize the class of all such tests by showing that any invariant decision statistic can be written as a function of two basic statistics which are in fact the adaptive matched filter (AMF) statistic and Kelly's generalized likelihood ratio statistic. Further, we establish an optimum test in the limit of low signal-to-noise ratio (SNR), the locally most powerful invariant (LMPI) test. We also derive the bound for the probability of detection of any invariant detector, at a fixed false-alarm rate, and compare the LMPI and the published detectors (Kelly and AMF) to it.

Journal ArticleDOI
TL;DR: Noise recorded inside the incubators in a neonatal intensive care unit and identified its sources reveal the need to train health care personnel on how to reduce such noise by taking more care in the handling of infants.
Abstract: This study evaluated the noise level inside the incubators in a neonatal intensive care unit and identified its sources in order to attempt to reduce it. Although noise is not a proven risk factor as far as the sensory integrity of newborns is concerned, it is certainly an important cause of stress to them and a source of serious and dangerous changes in their behavioral and physiologic states. Noise recorded inside the incubators had two components. The first was background noise from the incubator motors, which varied from 74.2 to 79.9 dB, and was similar to environmental noise. The second source was impulsive events beyond 80 dB. These events were the result of voluntary and involuntary contact with the incubators' Plexiglas surface or to the abrupt opening and closing of their access ports. Considering its decibel levels and frequency, this latter component is undoubtedly an important source of stress to newborns. Moreover, these data reveal the need to train health care personnel on how to reduce such noise by taking more care in the handling of infants.

Journal ArticleDOI
TL;DR: The findings suggest that AN preservation of AM coding in the presence of a continuous masking noise results from shifts in the operating ranges and firing rates of AN fibers resulting from cochlear nonlinearities and adaptive mechanisms.
Abstract: Sound envelope temporal fluctuations are important for effective processing of biologically relevant acoustic information including speech, animal vocalizations, sound-source location, and pitch. Amplitude modulation (AM) of sound envelopes can be encoded in quiet with high fidelity by many auditory neurons including those of the auditory nerve (AN) and cochlear nucleus. From both neurophysiological and clinical perspectives, it is critical to understand the effects of background masking noise on the processing of AM. To further this goal, single-unit recordings were made from AN fibers in anesthetized chinchillas. Units were classified according to spontaneous firing rate (SR) and threshold. Best frequency (BF) pure-tone bursts and AM (10-500 Hz) tone bursts were employed as stimuli at several sound levels, both in quiet and in the presence of a continuous wideband noise. It was found that (1) in quiet, low SR AN fibers show the strongest AM coding, followed in order by medium SR and high SR fibers, respectively. (2) AN units of all three classes generally preserve their AM coding even in the presence of loud (0 or +6 dB S/N) background noise and at high sound levels (over 75 dB SPL). (3) This preservation is usually achieved by lowering the average firing rate proportionately to decreases in the synchronous (fundamental frequency) response. (4) For a few AN fibers, the AM coding increases or is reduced in the presence of the background noise. These findings suggest that AN preservation of AM coding in the presence of a continuous masking noise results from shifts in the operating ranges and firing rates of AN fibers resulting from cochlear nonlinearities and adaptive mechanisms.

Journal ArticleDOI
01 Oct 1996
TL;DR: In this article, a canonical correlation analysis is applied to the outputs of two spatially separated arrays to detect the number of signals by testing the significance of the corresponding sample canonical correlation coefficients.
Abstract: A new approach is presented to the array signal processing problem of detecting the number of incident signals in unknown coloured noise environments with banded covariance structure. The principle of canonical correlation analysis is applied to the outputs of two spatially separated arrays. The number of signals is determined by testing the significance of the corresponding sample canonical correlation coefficients. The new method is shown to work well in unknown coloured noise situations and does not require any subjective threshold setting. The medium/high-SNR error rate may be approximately specified at a certain prescribed level, and may be traded off against the detection performance characteristic at low SNR. Simulation results are included to illustrate the performance of the proposed canonical correlation technique (CCT). It is found that the method performs well in a wide variety of coloured background noise environments. It is also demonstrated that the method is robust in the case when the noise covariance is not truly banded.

Journal ArticleDOI
TL;DR: Noise in the CSICU was above the recommended levels for patients and staff well-being and future studies will be designed to establish a correlation between sound levels and events.

Journal ArticleDOI
TL;DR: The basic technique, theoretically analyzes the algorithm to prove convergence under infinite time-average conditions, and demonstrates the algorithm via computer simulation for a single DSSS signal received in the presence of white Gaussian noise.
Abstract: A new technique for soft synchronization of direct-sequence spread-spectrum (DSSS) signals is presented. The technique, referred to as the dominant mode despreading (DMDS) algorithm, exploits the eigenstructure of a frequency-channelized DSSS signal to estimate the spreading code and underlying message sequence of the signal. Unlike other despreading techniques, the estimate of the code and data improves steadily with the number of code repeats. The technique is applicable to arbitrary spreading codes and message sequences and can operate in environments containing arbitrary levels of white background noise, and for signals with arbitrary unknown timing phase or carrier frequency offset. The technique requires the DSSS signal to have a constant-modulus spreading code and unrelated message and code-repeat rates. This paper introduces the basic technique, theoretically analyzes the algorithm to prove convergence under infinite time-average conditions, and demonstrates the algorithm via computer simulation for a single DSSS signal received in the presence of white Gaussian noise.

Patent
Jyri Suvanen1, Kirla Olli1
07 Jun 1996
TL;DR: In this article, an echo suppressor is placed on the side of the mobile network for eliminating the acoustic residual echo of the echo canceller of a mobile station in a mobile communications system.
Abstract: The invention relates to a method and arrangement for eliminating acoustic echo generated in a mobile station in a mobile communications system. According to the invention, an echo suppressor is placed on the side of the mobile network for eliminating the acoustic residual echo of the echo canceller of the mobile station. The echo suppressor comprises a downlink voice activity detector (35) because it is possible that the downlink speech returns from the mobile station as an acoustic echo superimposed to the uplink signal. When the detector (35) detects the downlink speech, a selector (303) disconnects the uplink signal from the speech decoder (38) and supplies in its place noise from the generator (302). The spectral characteristics and intensity of the comfort noise are similar to those of the background noise in the operating environment of the mobile station at each moment. Generation of the noise is started after a predetermined delay from detecting the voice activity, and terminated after a predetermined delay from the end of the voice activity.

Patent
01 Nov 1996
TL;DR: In this article, a pager or a cell phone includes an ambient condition sensor (13, 15, 29), and characteristics of an alert signal provided to inform a user of receipt of a message are altered according to sensed ambient conditions.
Abstract: A personal electronic device, such as a pager or a cell phone, includes an ambient condition sensor (13, 15, 29), and characteristics of an alert signal provided to inform a user of receipt of a message are altered according to sensed ambient conditions. Volume of an audio signal may be raised in the event of a higher sensed level of background noise, or pitch may be increased. In the case of light emitting alerts, the light output may be flashed, or intensity or color may be altered. In one embodiment a real-time clock (35) is incorporated, and alert characteristics are altered depending upon the time of day that a message is received. Some embodiments allow a user to program threshold conditions and direction of alteration of signal characteristics.