Showing papers on "Noise published in 1998"

Effects of noise and spectral resolution on vowel and consonant recognition: acoustic and electric hearing.

Abstract: Current multichannel cochlear implant devices provide high levels of speech performance in quiet. However, performance deteriorates rapidly with increasing levels of background noise. The goal of this study was to investigate whether the noise susceptibility of cochlear implant users is primarily due to the loss of fine spectral information. Recognition of vowels and consonants was measured as a function of signal-to-noise ratio in four normal-hearing listeners in conditions simulating cochlear implants with both CIS and SPEAK-like strategies. Six conditions were evaluated: 3-, 4-, 8-, and 16-band processors (CIS-like), a 6/20 band processor (SPEAK-like), and unprocessed speech. Recognition scores for vowels and consonants decreased as the S/N level worsened in all conditions, as expected. Phoneme recognition threshold (PRT) was defined as the S/N at which the recognition score fell to 50% of its level in quiet. The unprocessed speech had the best PRT, which worsened as the number of bands decreased. Recognition of vowels and consonants was further measured in three Nucleus-22 cochlear implant users using either their normal SPEAK speech processor or a custom processor with a four-channel CIS strategy. The best cochlear implant user showed similar performance with the CIS strategy in quiet and in noise to that of normal-hearing listeners when listening to correspondingly spectrally degraded speech. These findings suggest that the noise susceptibility of cochlear implant users is at least partly due to the loss of spectral resolution. Efforts to improve the effective number of spectral information channels should improve implant performance in noise.

The recognition of sentences in noise by normal-hearing listeners using simulations of cochlear-implant signal processors with 6–20 channels

Abstract: Sentences were processed through simulations of cochlear-implant signal processors with 6, 8, 12, 16, and 20 channels and were presented to normal-hearing listeners at +2 db S/N and at −2 db S/N. The signal-processing operations included bandpass filtering, rectification, and smoothing of the signal in each band, estimation of the rms energy of the signal in each band (computed every 4 ms), and generation of sinusoids with frequencies equal to the center frequencies of the bands and amplitudes equal to the rms levels in each band. The sinusoids were summed and presented to listeners for identification. At issue was the number of channels necessary to reach maximum performance on tests of sentence understanding. At +2 dB S/N, the performance maximum was reached with 12 channels of stimulation. At −2 dB S/N, the performance maximum was reached with 20 channels of stimulation. These results, in combination with the outcome that in quiet, asymptotic performance is reached with five channels of stimulation, demonstrate that more channels are needed in noise than in quiet to reach a high level of sentence understanding and that, as the S/N becomes poorer, more channels are needed to achieve a given level of performance.

Audio coding systems and methods

Abstract: An audio signal is decomposed into lower and upper sub-band and at least the noise component of the upper sub-band is encoded. At the decoder the audio signal is synthesised by a decoding means which utilises a synthesised noise excitation signal and a filter to reproduce the noise component in the upper sub-band.

Is level irrelevant in "irrelevant speech"? Effects of loudness, signal-to-noise ratio, and binaural unmasking.

Abstract: A series of experiments explored the role of level, signal-to-noise ratio, and the masking-level difference in the irrelevant speech effect (ISE). In Experiment 1 the detrimental effects of irrelevant sound on serial recall were found to be the same whether the material (speech or music) was presented at a high (75 dB[A]) or low (60 dB[A]) overall level. In Experiment 2, adding pink noise to the speech signal produced a linear improvement in performance with decreasing speech-to-noise ratios. In Experiment 3 the contribution of binaural unmasking to the ISE was found to be negligible. The results (a) confirm that the segmented, changing nature of the irrelevant sound is crucial in producing the ISE and (b) suggest that the adverse effects of disruptive auditory input may be alleviated by introducing additional uniform masking noise.

Adaptive-neighborhood filtering of images corrupted by signal-dependent noise.

Abstract: In many image-processing applications the noise that corrupts the images is signal dependent, the most widely encountered types being multiplicative, Poisson, film-grain, and speckle noise. Their common feature is that the power of the noise is related to the brightness of the corrupted pixel. This results in brighter areas appearing to be noisier than darker areas. We propose a new adaptive-neighborhood approach to filtering images corrupted by signal-dependent noise. Instead of using fixed-size, fixed-shape neighborhoods, statistics of the noise and the signal are computed within variable-size, variable-shape neighborhoods that are grown for every pixel to contain only pixels that belong to the same object. Results of adaptive-neighborhood filtering are compared with those given by two local-statistics-based filters (the refined Lee filter and the noise-updating repeated Wiener filter), both in terms of subjective and objective measures. The adaptive-neighborhood approach provides better noise suppression as indicated by lower mean-squared errors as well as better retention of edge sharpness than the other approaches considered.

Postprocessing method for suppressing musical noise generated by spectral subtraction

Abstract: We investigate whether musical noise, which often exists in speech enhanced using spectral subtraction, can be suppressed. Via exploiting some specific characteristics of human speech, we propose a method that can effectively suppress musical noise without a noticeable effect on speech intelligibility. Performance assessments confirm that our method is effective.

A novel psychoacoustically motivated audio enhancement algorithm preserving background noise characteristics

Abstract: In this paper we propose an algorithm for reduction of noise in audio signals. In contrast to several previous approaches we do not try to achieve a complete removal of the noise, but instead our goal is to preserve a pre-defined amount of the original noise in the processed signal. This is accomplished by exploiting the masking properties of the human auditory system. The speech and noise distortions are considered separately. The spectral weighting rule, adapted by utilizing only estimates of the masking threshold and the noise power spectral density, has been designed to guarantee complete masking of distortions of the residual noise. Simulation results confirm that no audible artifacts are left in the processed signal, while speech distortions are comparable to those caused by conventional noise reduction techniques.

Apparatus and method of in-service audio/video synchronization testing

Abstract: An apparatus and method provide non-intrusive in-service testing of audio/video synchronization testing without using traditional audio marker tones. The network includes an A/V synchronous test signal generator which injects video and audio markers into the video and audio non-intrusively and routes the two signals into a switch where they are switched into a channel for encoding and transmission via the ATM network. At the distant end the signal is decoded and routed by a switch into the A/V test generator and measurement set where the markers are detected and the A/V skew calculated, after which the audio and video are routed to the subscriber. The A/V test set signal generator includes a Video Blanking Interval (VBI) test signal generator and a white noise generator, the former injecting a marker into the video signal and the later injecting an audio marker into the audio signal. The video marker is injected into the VBI and broadband, background audio noise to measure the delay between the audio and video components of a broadcast. The marking of the audio is accomplished by gradually injecting white noise into the audio channel until the noise level is 6 dB above the noise floor of the audio receiver. As a precursor A/V sync signal, a small spectrum of the white noise is notched or removed. This signature precludes inadvertent recognition of program audio noise as the audio marker.

Masked hearing thresholds of a beluga whale (Delphinapterus leucas) in icebreaker noise

Abstract: An experiment is presented that measured masked hearing thresholds of a beluga whale at the Vancouver Aquarium. The masked signal was a typical beluga vocalization; the masking noise included two types of icebreaker noise and naturally occurring icecracking noise. Thresholds were measured behaviorally in a go/no-go paradigm. Results were that bubbler system noise exhibited the strongest masking effect with a critical noise-to-signal ratio of 15.4 dB. Propeller cavitation noise completely masked the vocalization for noise-to-signal ratios greater than 18.0 dB. Natural icecracking noise showed the least interference with a threshold at 29.0 dB. A psychophysical analysis indicated that the whale did not have a consistent decision bias.

Noise reduction by paired-microphones using spectral subtraction

Abstract: This paper proposes a method of noise reduction by paired microphones as a front-end processor for speech recognition systems. This method estimates noise using a subtractive microphone array and subtracts them from the noisy speech signal using spectral subtraction (SS). Since this method can estimate noise analytically and frame by frame, it is easy to estimate noise not depending on these acoustic properties. Therefore, this method can also reduce non-stationary noise, for example sudden noise when a door has just closed, which cannot be reduced by other SS methods. The results of computer simulations and experiments in a real environment show that this method can reduce LPC log spectral envelope distortions.

Quick audio retrieval using active search

Abstract: This paper discusses a method to search quickly through broadcast audio data to detect and locate known sounds using reference templates, based on the active search algorithm and histogram modeling of zero-crossing features. Active search reduces the number of candidate matches between reference and test template by up to 36 times compared to exhaustive search, while still remaining optimal. Computation is further reduced by using computationally inexpensive zero-crossing features. The method is robust against white noise addition down to 20 dB signal-to-noise ratios and digitization noise.

An efficient method for the removal of impulse noise from speech and audio signals

Abstract: A computationally efficient algorithm is proposed to remove noise impulses from speech and audio signals while retaining its features and tonal quality. The proposed method is based on the SD-ROM (Signal Dependent Rank Order Mean) algorithm. This technique has successfully been used to remove impulse noise from images. It has the advantage of being relatively fast, simple and robust. The algorithm estimates the likelihood the sample under inspection is corrupt relative to the neighboring samples and replaces a sample detected as corrupted by a value based on the neighboring samples. This algorithm also has the advantage of being 'configurable' to the type of noise impulses in the sample, as the thresholds used to detect noise impulses can be varied to suit the signal.

Submicrovolt flicker electroretinogram : Cycle-by-cycle recording of multiple harmonics with statistical estimation of measurement uncertainty

Abstract: Purpose To study cycle-by-cycle recording of small-amplitude flicker-electroretinogram (ERG) responses and analyze results with robust statistical methods to estimate the measurement uncertainty. Methods Flicker ERGs at 32 Hz were recorded simultaneously from both eyes of patients with retinal degeneration. The ERG was amplified under wide-band (1-1000 Hz) conditions, digitized at 6144 Hz/eye, and multiplied point for point (192 points/cycle) by sine and cosine functions within each 1/32-second flash cycle to extract coefficients for six harmonic components of a discrete Fourier transform in real time. Amplitude windowing was not used, and all data were saved for subsequent statistical processing to identify and remove large-amplitude artifacts discretely and to search for quiet recording periods that minimized small-amplitude noise. Results Plots of amplitude and phase indicated far outlying noise points that were excised from the data. The SD of sequential intervals on a time line of the sine component identified quiet periods that minimized small-amplitude noise and improved measurement consistency. The SE of the response mean provided an estimate of measurement uncertainty. Conclusions The harmonic components of many individual responses are captured quickly (e.g., 500 responses in 15.6 seconds) for post hoc statistical analysis, using mathematical algorithms that are precisely reproducible to facilitate comparison of results from all laboratories. Graphical time lines of responses allow separation of artifact transients from gaussian noise for elimination of noisy periods without disturbing the stored information. Statistical estimates of measurement uncertainty are determined on-line to allow immediate feedback during the recording session. Amplitude-phase plots of the multiple harmonic components, along with reconstructed analog waveforms, provide results in a readily assimilated manner for comparison of all testing sessions.

Multichannel noise reduction — Algorithms and theoretical limits

Abstract: In this contribution we present a theoretical study on the performance of several multichannel noise reduction algorithms. It is known that the magnitude squared coherence (MSC) determines the performance of a class of adaptive algorithms i.e. active noise control or noise reduction with a reference microphone. However, the MSC is not sufficient for performance evaluation of other noise reduction methods like the Generalized Sidelobe Canceller (GSC) or adaptive post-filter techniques. We propose to measure the spatial complex coherence (CC) or normalized cross power spectrum of the sound field and show that it can be used as a much more general tool for performance analysis. First of all, we summarize the results of previous studies and present new results for the performance of the Generalize Sidelobe Canceller (GSC) as a function of the complex coherence. In the second part we examine different noise fields to show theoretical limits of multichannel noise reduction schemes.

Engine noise simulating novelty device

Abstract: An engine noise simulating novelty device is provided including a speaker for audibly transmitting audio signals upon the receipt thereof. Further included is a sound module connected to the speaker and a tachometer of a vehicle. The sound module is adapted to communicate audio signals with the speaker which represent a sound, wherein a frequency of the sound is varied with a change in the revolutions per minute of the engine of the vehicle, as indicated by the tachometer.

Audio information sorter

Abstract: PROBLEM TO BE SOLVED: To make easily and quickly determinable between silent and sound blocks, and make sortable to music blocks and voice blocks, or to music blocks, voice blocks and noise blocks. SOLUTION: A sound determining part 4 determines the existence of sound when a value σe2 found by an energy dispersion operating part 3 is larger than a threshold value. When the existence of the sound is determined, audio information stored in a memory 5 is read out to be input to a sum-of-whole-subband- energy operating part 7 and a gravity-center-of-subband-energy operating part 11. Outputs of the operating part 7 are binarized by a binarization operating part 8 to find dispersion σs2 within a unit time of a binarized series in a binarized series dispersion operating means 9. On the other hand, A mean.dispersion-of-energy-gravity-center operating part 12 finds a mean of energy gravity centers Eg and dispersion σg2. An audio information identification part 10 conducts determination of noises, music and sound relating to the dispersion σs2, the mean Eg and dispersion σg2, using an identification function. COPYRIGHT: (C)2000,JPO

Method and apparatus for characterizing audio transmitting system, and method and apparatus for setting characteristics of audio filter

Abstract: A designing system for adaptively characterizing an audio transmitting system has a white noise generating unit for generating a white noise signal. A speaker radiates the white noise generated by the white noise generating unit into an acoustic space. A microphone is placed at a predetermined position in the acoustic space and collects sound radiated from the speaker. A FIR adaptive filter receives the above white noise signal. An LMS algorithm processing unit updates each tap coefficient of the adaptive filter by using the LMS algorithm. A computation unit calculates the difference between a detection signal output from the microphone and an output of the adaptive filter and outputs the difference as an error signal e. By using a white noise signal having an average power of one, the range of the step size parameter of the LMS algorithm required for stably operating the adaptive filter is fixed. According to the method for characterizing the audio transmitting system and the method for setting an audio filter, the stable operation of the adaptive filters is ensured without increasing the complexity of the overall system.

Noise reduction scheme for a computer system

Abstract: A noise reduction scheme is incorporated into a computer system for reducing environmental background noise when a user is listening to audio output through a standard set of headphones. The noise reduction scheme utilizes the processing power of the computer system as well as built-in components of the computer for enhancing the audio quality heard by the user. Environmental background noise is received by a built-in microphone, wherein a reciprocal noise canceling signal is generated by the computer's microprocessor. The reciprocal noise canceling signal is then mixed with audio from the computer system for reducing the environmental background noise heard by a user.

Theoretical study of the recording density limit of a near-field photochromic memory

Abstract: The recording density limit of a near-field optical memory that uses a photochromic medium was theoretically studied by use of Shannon’s information theory. Shot noise and material noise were taken into account in the analysis of the signal-to-noise ratio. The conventional recording density limit, which is defined by the inverse of the minimum recorded mark area, and Shannon’s recording density limit were evaluated. The conventional recording density limit was 1011–1012 bits/cm2, and Shannon’s recording density limit was 1012–1013 bits/cm2.

Speech enhancement for mobile telephony based on non-uniformly spaced frequency resolution

Abstract: In this paper, we present a speech enhancement method based on spectral subtraction in non-uniformly spaced sub-bands. The main advantage of using different frequency resolutions for the various bands is the perception property of the human ear, which is able to separate low frequencies more precisely than high frequencies. The parameters of the noise reduction algorithm are chosen appropriately to the respective signal to noise ratio, which yields a performance superior to an uniform frequency resolution with the same number of sub-bands. A two-stage cascaded filter-bank is used for the decomposition of the signal.

Noise making novelty baseball bat

Abstract: A souvenir or keepsake baseball bat for baseball fans including young children. The souvenir baseball bat is approximately one-third the size of an actual baseball bat in length and dimension and includes a hollowed out portion that is filled with beans, beads, buckshot or the like which causes the souvenir bat to rattle when waved or shaken.

Rough Set Methods and Applications

Abstract: As is shown by the results of some recently proposed methods, the perceptual coding of audio can be used for suppressing the noise affecting transmitted audio signals The process of tuning the perceptual audio coding algorithm demands finding the relations between masking algorithm parameters and their influence on the subjective quality of processed audio The rough set method was employed to discover ill-defined relations underlying the implemented perceptual model of hearing

Adding noise during LPC coded voice activity periods to improve the quality of coded speech coexisting with background noise

Abstract: A system and method to improve the quality of coded speech coexisting with background noise. For instance, the present invention receives a coded speech signal via a communication network and then decodes and synthesizes the different parameters contained within it to produce a synthesized speech signal. The present invention determines the non-speech periods that are represented within the synthesized speech signal. The determined non-speech periods are then utilized to determine and code LPC parameters needed for background noise synthesis. Because medium or low bit rate LPC-coded speech during voice activity periods has the coexisting background noise attenuated, the decoded signal has audible abrupt changes in the level of the background noise. To improve decoded speech quality, the present invention adds simulated background noise to decoded noisy speech when synthesizing the noisy speech signal during voice activity periods. The resulting output signal sounds more natural and realistic to the human ear because of the continuous presence of background noise during speech and non-speech periods.

Radio receiver squelch circuit with automatic adjustment of the squelch threshold

Abstract: A radio receiver for receiving audio frequency communications conveyed by a high-frequency carrier and occurring randomly over time. In addition to the tuning circuits, a director, an audio frequency amplifier and a speaker, the receiver includes a controlled switch connected between the detector and the amplifier for disabling or re-enabling the transfer of the audio signal from the detector to the speaker. A filter and second detector set a silence threshold to provide squelch and are arranged to set the silence threshold in accordance with the level of a part of the audio spectrum from the first named detector that contains only noise and is located outside the frequency band of the signal representing the useful audio communication to be received. The receiver is particularly suitable for use in citizen's band equipment.

A geometric algorithm for voice activity detection in nonstationary Gaussian noise

Abstract: A new algorithm for voice activity detection in additive nonstationary noise is presented The algorithm utilizes the differences of the probability distribution properties of noise and speech signal The Magnitude Density (MDF) and the Magnitude Distribution Functions (MDF) are defined The noise level is monitored for automatic threshold estimation The estimate is shown to be accurate also when analysis windows do not fully contain non-speech signals and in the presence of non-stationary noise The algorithm has been applied different type of noises (traffic, water, restaurant, ect) The voice activity detection algorithm is shown to operate reliably in SNRs down to 0 dB and noise variance up to 10 dB/sec

Robust features derived from temporal trajectory filtering for speech recognition under the corruption of additive and convolutional noises

Abstract: This paper presents a novel method using robust features for speech recognition when the speech signal is corrupted by additive and convolutional noises. This method is conceptually simple and easy to be implemented. The additive noise and the convolutional noise are removed by temporal trajectory filtering in the autocorrelation domain and cepstral domain, respectively. No prior information of noise corruption is necessary. A task of multi-speaker isolated digit recognition is conducted to demonstrate the effectiveness of using these robust features. The cases of the channel filtered speech signal corrupted by additive white noise and color noise are tested. Experimental results show that significant improvements can be achieved as compared with some traditional features.

Time stability characterization and spectral aliasing. Part II: a frequency-domain approach

Abstract: We showed in Part I of this paper that the estimation of the white-phase noise level of a sampled signal may be achieved with variances even if the sampling frequency is far lower than the high cut-off frequency. In Part II, the effects of spectral aliasing for the different types of noise are reviewed. The responses of the Allan variance and the modified Allan variance for high-frequency noises are calculated taking into account the spectral aliasing. It is demonstrated that the effects of spectral aliasing for low-frequency noises may be neglected.

Combined residual echo and noise reduction: A novel psychoacoustically motivated algorithm

Abstract: In this paper we focus on the problem of acoustic echo cancellation and noise reduction for hands-free telephony devices. A standard echo canceller is combined with a frequency domain post-filter, which applies a novel psychoacoustically motivated weighting rule. The algorithm makes use of the masking threshold of the human auditory system to achieve a perceived reduction of noise and residual echo equal to some pre-defined levels. In contrast to conventional methods, the proposed one preserves the nature of the original background noise and doesn't introduce any audible artifacts. At the same time it can attain a very high reduction of the residual echo.