Showing papers on "Noise published in 1996"

Speech signal transmitting and receiving apparatus with noise sensitive volume control

Abstract: A speech signal transmitting receiving apparatus, such as a portable telephone set, includes a speech signal transmitting encoding circuit, a noise domain detection unit, a noise level detection unit and a controller. The speech signal transmitting encoding circuit compresses input speech signals by digital signal processing at a high efficiency. The noise domain detection unit detects the noise domain using an analytic pattern produced by the speech signal transmitting encoding circuit. The noise level detection unit detects the noise level of the noise domain detected by the noise domain detection unit. The controller controls the received sound volume responsive to the noise level detected by the noise level detection unit.

Derivative analysis of hyperspectral data

Abstract: With the goal of applying derivative spectral analysis to analyze high resolution, spectrally continuous remote sensing data, several smoothing and derivative computation algorithms have been reviewed and modified to develop a set of cross-platform spectral analysis tools. Emphasis was placed on exploring different smoothing and derivative algorithms to extract subtle spectral features from any continuous spectral data sets. With interactive selection of bandwidth and sampling interval (band separation), the algorithm can optimize noise reduction and better match the scale of spectral features of interest. Laboratory spectral data were used to test the performance of the implemented derivative analysis modules. An algorithm for detecting the absorption band positions was executed on synthetic spectra and a soybean fluorescence spectrum to demonstrate the usage of the implemented modules in extracting spectral features. Upon examination of the developed modules, issues related to the smoothing and the spectral deviation caused by the smoothing or derivative computation algorithms were also observed and discussed. The scaling effect resulting from the migration of band separations when using the finite approximation derivative algorithm was thoroughly inspected to understand the relationship between the scaling effect and noise removal.© (1996) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Automatic volume adjusting device and method

Abstract: A method and device for automatically adjusting the volume of an existing audio device such as a television and/or stereo receiver. The automatic volume adjusting device of the present invention includes a receiver for receiving control signals transmitted from a remote control transmitter associated with the audio device, a transmitter for transmitting control signals to a control signal receiver of the audio device, a microphone for sensing an ambient noise level, and a controller coupled to the receiver, to the transmitter, and to the microphone. The controller operates in a training mode and an operating mode. In the training mode, the controller learns codes associated with volume adjusting control signals transmitted by the remote control transmitter. In the operating mode, the controller determines when the ambient level of noise detected by the microphone is outside a predefined volume range and adjusts the volume of the remotely controlled audio device to fall within the predefined range by transmitting the learned control signals to the audio device. The automatic volume adjusting device may further include a push-button toggle switch and an indicator light, such as a light emitting diode (LED) for placing the controller in a training or operating mode and a power converter for converting power supplied from a 110-volt AC wall outlet to a DC voltage for internal use within the device.

Integrating audio and visual information to provide highly robust speech recognition

Abstract: There is a requirement in many human machine interactions to provide accurate automatic speech recognition in the presence of high levels of interfering noise. The the paper shows that performance improvements in recognition accuracy can be obtained by including data derived from a speaker's lip images. We describe the combination of the audio and visual data in the construction of composite feature vectors and a hidden Markov model structure which allows for asynchrony between the audio and visual components. These ideas are applied to a speaker dependent recognition task involving a small vocabulary and subject to interfering noise. The recognition results obtained using composite vectors and cross-product models are compared with those based on an audio-only feature vector. The benefit of this approach is shown to be an increased performance over a very wide range of noise levels.

Residual modeling in music analysis-synthesis

Abstract: In analysis-synthesis of musical sounds based on a sinusoidal model, the difference between the original signal and the synthesized signal, termed the residual, is typically a broadband noise process. It contains such musical phenomena as flute breath noise or violin bow noise. Synthesis without such "noise" tends to sound artificial; it is desirable to improve the synthesis realism by modeling the residual in such a way that it can be reinjected in the synthesized signal. This paper deals with a model of noise perception based on the equivalent rectangular bands (ERBs) of the auditory system. Since a broadband noise is perceptually well-represented by the time-varying energy in each of these frequency bands, the residual is parametrized in terms of these energies in the proposed model. An application of the model to music synthesis based on the inverse fast Fourier transform (FFT) is described in detail.

Combination of two-channel spectral subtraction and adaptive wiener post-filtering for noise reduction and dereverberation

Abstract: In this contribution a novel structure for the enhancement of speech signals disturbed by acoustic noise is presented which is based on Spectral Subtraction. The Spectral Subtraction technique is combined with a novel estimator for the noise power spectrum which takes advantage of the employment of a second microphone. Due to the extension to a two-microphone system the Spectral Subtraction can be used to reduce realistic, non-stationary noise sources. Additionally, the performance of the system is further improved by the application of a post filter adapted according to Wiener filter techniques. As a result, the proposed speech enhancement system provides a significant suppression of noise in realistic situations as well as a reduction of room reverberation.

A guide to writing as an engineer

Abstract: Engineers and Writing. Some Guidelines for Good Engineering Writing. Eliminating Sporadic Noise in Writing. Writing Letters, Memoranda, and Electronic Mail. Writing Some Common Engineering Documents. Writing an Engineering Report. Accessing Engineering Information. Engineering Your Presentations. Writing to Get an Engineering Job. Writing with Computers.

Reduced frequency selectivity and the preservation of spectral contrast in noise.

Abstract: Reduced frequency selectivity associated with sensorineural hearing loss may pose particular problems for hearing-impaired listeners in noisy environments. In these situations, broader-than-normal auditory filters may affect the perception of speech by reducing the contrast between spectral peaks and valleys in at least two ways. First, the peaks and valleys in the internal representation of the speech spectrum become smeared, resulting in less precise frequency analysis. Second, there may be a reduction in the signal-to-noise ratio (S/N) at the output of each auditory filter. In order to examine the relationship between frequency selectivity and identification of speechlike stimuli in noise, hearing-impaired and normal-hearing listeners were trained to assign vowel labels to four harmonic complexes which differed in the frequency locations of four elevated ("peak") harmonics. Peak harmonics were chosen to approximate first- and second-formant frequencies in four English vowels. Listeners were then tested to determine the spectral contrast necessary between peak and background components in order to maintain identification accuracy in the presence of various levels of broadband noise. Results indicated that for these stimuli, normal-hearing listeners required about 1 dB of additional spectral contrast for every doubling of the intensity of noise. The required increase in spectral contrast was generally greater for listeners with broader-than-normal auditory filters at 2000 Hz. This finding suggests indirectly that in the internal representations of speech sounds embedded in noise, the signal-to-noise ratio for listeners with abnormal frequency selectivity is poorer than for listeners with normal frequency selectivity. A poorer-than-normal internal S/N may be one factor underlying the common observation that noise often is more degrading to speech understanding by hearing-impaired listeners than by normal-hearing listeners.

Voice activity detector for half-duplex audio communication system

Abstract: A method (501, 502) of detecting voice in an audio signal comprises the steps of determining an average peak value (703, AP) representing an envelope of the audio signal, determining a running instance of audio signal standard deviation (702), which corresponds to one of a number of overlapping time intervals, and updating a power density function (PDF) by adding instances of noise to the PDF if the average peak of the audio signal exceeds the current level of the audio signal by a certain amount and if the current standard deviation value falls below a threshold for a predetermined time interval. A noise floor (NF) is located based on the mean value of the PDF (501), and, if the audio signal sustains a power level exceeding the noise floor, voice activity is determined to be present in the audio signal (502). The PDF is updated by a low confidence factor (1206) if all of the standard deviation values calculated during a certain period of time are below the threshold value and by a high confidence factor (1204) if all standard deviation values within a certain longer period of time period are below the threshold value.

Display of geographic locations with correlated signal quality measurements

Abstract: A visual presentation of a simulated communication unit moving through a wireless communication system is combined with an audio presentation of speech that a user would hear (311) if the user were placed in the same radio environment. In addition, both audible and visual indicia of RF interference may also be provided (107, 109), with or without audio playback, to aid in designing around interference sources and their intensities. A set of error masks and corresponding audio files are created (305), wherein each file contains the same audio track, but based on speech error masks created at different vehicle speeds and signal to noise ranges. Real-time playback of the audible signal with noise is useful in planning for particular needs of a wireless communication system planner.

Method and apparatus for multi-channel active control of noise or vibration or of multi-channel separation of a signal from a noisy environment

Abstract: The present invention is directed to a system and method for feed-forward active control of noise and vibration. In operation, at least one of noise and vibration from potential noise and vibration sources are detected, and noise reference data based on the detection of noise and vibration from the potential noise and vibration sources is generated. Further, at least one of noise and vibration at a selected environment in which noise and vibration are to be minimized are also detected, whereby error data based on the detection of noise and vibrations at the selected environment is generated. Filter constants are generated based on the noise reference data and the error data, wherein the generating of the filter constants includes the elimination of redundancies in the noise reference data. The noise reference data is processed based on the generated filter constants, whereby noise/vibration canceling outputs based on the processed noise reference data is generated to minimize energy of the noise and vibration detected at the selected environment.

Extended spectral subtraction

Abstract: This paper describes a new method for one channel noise suppression system which overcomes the typical disadvantage of one channel noise suppression algorithms — the impossibility of noise estimation during speech sequence. Our method is the combination of Wiener filtering and spectral subtraction. The noise can be successfully updated even during the speech sequences and that is why there is no need of the voice activity detector.

Method and apparatus for improving effective signal to noise ratios in hearing aids and other communication systems used in noisy environments without loss of spectral information

Abstract: Improved signal to noise ratio to help speech comprehension in a noisy environment is accomplished by selectively downwardly expanding a speech and noise composite signal when the speech signal is absent, thereby lowering signal components which represent noise. An expansion control signal is extracted from the composite input signal. Operation is based on the assumption that when noise alone is present, the input signal amplitude is less than some reference level and that when speech and noise are present together, the input signal amplitude is greater than the reference level. The response rates of gain changes are quite rapid, and do not introduce distortion or other audibly noticeable artifacts of the processing. The amount of downward expansion of the noise alone is small compared to noise gates to further reduce processing artifacts. The methods of realization include use of, in combination and alone, analog compressors and expanders, analog expanders in combination with voltage clamps and/or automatic level control circuits, two-quadrant multipliers in conjunction with digital control, entirely digital means for obtaining the requisite sensing and gain control, and expandor designs that are analogs of conventional filter designs where the notion of amplitude replaces frequency. Automatic noise suppression may be employed to pre-process the input signal, thereby rendering the control circuit self adjusting for better performance over a wide range of background sound levels. Various microphone, preferably providing directional characteristics, may be used to reduce noise levels in the received input signal.

Method and apparatus for reducing noise in speech signal

Abstract: A method and an apparatus for reducing the noise in a speech signal capable of suppressing the noise in the input signal and simplifying the processing. The apparatus includes a fast Fourier transform unit 3 for transforming the input speech signal into a frequency-domain signal, and an Hn value calculation unit 7 for controlling filter characteristics for filtering employed for removing the noise from the input speech signal. The apparatus also includes a spectrum correction unit 10 for reducing the input speech signal by the filtering conforming to the filter characteristics produced by the Hn value calculation unit 7. The Hn value calculation unit 7 calculates the Hn value responsive to a value derived from the frame-based maximum SN ratio of the input signal spectrum obtained by the fast Fourier transform unit 3 and an estimated noise level and controls the processing for removing the noise in the spectrum correction unit 10 responsive to the Hn value.

Electronic cancellation of ambient noise in telephone headset

Abstract: Noise reducing circuits for electronic receiving instruments, such as telephone receivers in headsets or handsets that are used in noisy locations, provide compensation of the set's receiver unit-to-error microphone transfer function to enhance the noise reduction. Further circuits provide pre-conditioning of the incoming voice signal to make the noise cancellation more effective. The tendency of these noise cancelling circuits to oscillate is substantially lessened by added circuitry which rapidly detects onset of oscillation and momentarily reduces the noise cancellation without interrupting the incoming speech path altogether.

Floor impact noise suppressor in a multi-storied building

Abstract: A noise suppressor for reducing propagation of a floor impact noise generated in an upper story to a lower story in a multi-storied building is disclosed. A reference sensor set at a position between a floor of the upper story and a ceiling of the lower story detects a floor impact noise generated in the upper story and converts the noise to an electric signal. This signal is computed and processed by a control unit and transmitted to a speaker located at a position lower than the reference sensor. The speaker emitts a sound wave interfering the floor impact noise to eliminate the noise. The noise not having been suppressed is detected by an error sensor and processed together with the noise detected by the reference sensor by the control unit, thus a sound wave to be emitted from the speaker is corrected appropriately.

Mining knowledge in Noisy audio data

Abstract: This paper demonstrates a KDD method applied to audio data analysis, particularly, it presents possibilities which result from replacing traditional methods of analysis and acoustic signal processing by KDD algorithms when restoring audio recordings affected by strong noise.

Robust treatment of impulsive noise in speech and audio signals

Abstract: Markov chain Monte Carlo methods are presented for treatment of localized, impulsive noise (outliers) in digitized waveforms, within a Bayesian hierarchical framework. Outliers in audio signals occur as`clicks' and`crackles' in degraded sound recordings and impulsive noise in communications channels. Sampling-based methods for detection and correction of such artefacts are presented, in which individual noise sources are modelled as Gaussian with unknown scale, allowing for robustness to heavy-tailed noise distributions. Results are presented for speech and audio signals obtained from digitized sound recordings.

Influence of Acquisition Parameters on the Measurement of Click Evoked Otoacoustic Emissions in Neonates in a Hospital Environment

Abstract: There is much interest in the introduction of a universal neonatal hearing screening programme. Screening programmes using high-risk criteria have been used for some time, but 50 per cent of deaf and hearing-impaired neonates are not identified because they are not classified as high risk for hearing impairment at birth. Otoacoustic emission (OAE) measurement is widely regarded as a technique likely to be suitable for universal hearing screening. To examine this, otoacoustic emissions were measured from 351 neonate ears at a large maternity hospital. Of particular interest were the practicalities of recording OAE in a hospital environment, the establishment of an appropriate age at which screening should be performed on neonates and investigation of the relative advantages of different recording techniques. Main findings were: (1) low OAE levels relative to noise during the first 24 to 48 hours post partum; (2) lower OAE signal to noise levels in the low frequencies irrespective of age; (3) increase of ov...

Audio signal processing method

Abstract: PROBLEM TO BE SOLVED: To always listen to music by the same frequency balance. SOLUTION: The frequency division into two bands or more is performed for an audio signal, the compressibility of each frequency band is determined based on a loudness characteristic and a compressor 4 compresses, synthesizes and outputs the audio signal of each frequency band for which the frequency division is performed based on the compressibility. Based on the relation of the size of the ruing noise according to the loudness characteristic and running speed and the frequency, the compressibility of each frequency band in the present running speed is determined and the audio signal of each frequency band for which the frequency division is performed is compressed, synthesized and outputted based on the compressibility. Further, based on the maximum value of listening sound by the loudness characteristic and volume, the compressibility of each frequency band is determined, and the audio signal of each frequency band for which the frequency division is performed is compressed, synthesized and outputted based on the compressibility. COPYRIGHT: (C)1997,JPO

Adaptive microphone arrays for speech enhancement in coherent and incoherent noise fields

Abstract: To enhance the quality of speech in a hands–free telecommunication environment, microphone arrays consisting of multiple microphones can be used for sound pick–up. To circumvent the limitations associated with small array apertures adaptive signal processing techniques can be used. For extreme situations of a free (coherent) and a diffuse (incoherent) noise field, acceptable solutions for the noise reduction problem exist: In coherent noise fields, the adaptive beamforming techniques yield a high‐noise reduction performance, but loose their efficiency in incoherent noise fields. On the other hand, for incoherent noise fields microphone arrays with adaptive post–filtering yield good performance; but in a coherent noise environment this method may cause unacceptable distortions in the output signal. In this contribution the advantages and disadvantages of these two techniques for noise reduction are discussed. A technique is presented which operates independently of the correlation properties of the noise fields, i.e., this method is able to reduce coherent and incoherent noise simultaneously. Some results are given using objective quality measures as well as subjective listening tests. The methods are also compared by using the recognition rate of a speaker‐independent isolated word recognition system.

Method and apparatus for reducing noise using a plurality of recording copies

Abstract: The present invention provides a method for reducing noise using a plurality of recording copies. The present invention produces a master file with lower noise than the available recording copies, and avoids the problems of losing musical content caused by prior art pop and click removers. The system comprises a recording playback unit, a computer system with a sound input capability, and a high capacity storage system such as a CD recorder. In operation, a plurality of recording copies of a single recording are played on the playback unit. These recordings are digitized by the computer and a separate recording file is formed for each copy of the recording. The recording files are then synchronized. The samples from each of the recording files are then averaged to reduce the noise components. A variety of threshold comparison techniques can be employed to eliminate samples and/or recording files that are outside of a computed range for that sample based on the values of the master, the other slave files or a combination thereof.

Audible in Utero Sound Caused by the Ultrasonic Radiation Force from a Real-Time Scanner

Abstract: While investigating in utero sound levels during vibro-acoustic stimulation on the maternal abdomen it was noticed that noise level increased when the real-time ultrasonic scanner beam was directed at the sensing hydrophone. The noise was recorded and later analysed for frequency content and waveform. It appeared related to the scanning and frame rate frequencies of the scanner used. Sounds may originate from radiation pressure produced when the ultrasound beam is absorbed by tissue or reflected from bone or the metal hydrophone. This implies that although ultrasound cannot be heard per se, any modulation of its intensity will produce vibrations in the maternal tissues or reflecting structures such as skull bone, and especially stapes, malleus and incus, that would be heard as sound by the fetus. The intensity of the sound produced varied with orientation of the transducer beam and this may itself produce a stimulation. Based on our recordings (Fig. 1), it was calculated (please see Appendix) that the fetus would hear a sound corresponding to 84dB noise pressure level in air.

Consonant Confusions in Amplitude-Expanded Speech

Abstract: The perceptual consequences of expanding the amplitude variations in speech were studied under conditions in which spectral information was obscured by signal correlated noise that had an envelope ...

A comparison of linear and nonlinear scale-space filters in noise

Abstract: The properties of two scale-space systems are compared by examining their performance in noise. It is found that in Gaussian noise linear diffusion and a new type of filter called the area sieve have similar performance but in impulsive noise of random amplitude the area sieve is superior.

Noise level in a care and teaching hospital institution

Abstract: Noise in the environment is increasing over the years. Disturbances produced by noise are varied, some lead to serious health consequences. Noise level was registered in a teaching hospital. Levels in the wards were between 50 and 59 dB. In the Intensive Care Unit, main hallways and outpatients department levels were higher than 59 dB. Isolated peaks up to 90.0 dB (Pediatrics) were detected. The noise level recommended for a hospital is under 50.0 dB. We found that the principal source of noise came from the medical and nursing staff.

Improved spectral subtraction for speech enhancement

Abstract: The spectral subtraction approach has become almost standard in speech enhancement because it is relatively easy to understand and implement The major drawback of the spectral subtraction method is that it leaves residual noise with annoying noticeable tonal characteristics referred to as musical noise For low SNR the perceived effect of the "musical noise" is close to that of the additive noise In the present work we propose to reduce the musical noise by applying the output of a standard spectral subtractor to a constrained high order notch filter which suppresses the "musical noise" The filtration process distorts the speech signal It is possible to reduce the level of distortion if the speech signal is preprocessed properly before it is contaminated by the noise It will be demonstrated that the proposed method is superior to the standard spectral subtraction specially for low SNR A comprehensive listening test indicated that for segmental SNR= −12dB, 77% of the listeners strongly preferred the proposed approach over the usual spectral subtraction approach