19419 results about "Noise reduction" patented technology

Apparatus, method, logic arrangement and storage medium are provided for increasing the sensitivity in the detection of optical coherence tomography and low coherence interferometry (“LCI”) signals by detecting a parallel set of spectral bands, each band being a unique combination of optical frequencies. The LCI broad bandwidth source can be split into N spectral bands. The N spectral bands can be individually detected and processed to provide an increase in the signal-to-noise ratio by a factor of N. Each spectral band may be detected by a separate photo detector and amplified. For each spectral band, the signal can be band p3 filtered around the signal band by analog electronics and digitized, or, alternatively, the signal may be digitized and band pass filtered in software. As a consequence, the shot noise contribution to the signal is likely reduced by a factor equal to the number of spectral bands, while the signal amplitude can remain the same. The reduction of the shot noise increases the dynamic range and sensitivity of the system.

A proximity sensor system includes a sensor matrix array having a characteristic capacitance on horizontal and vertical conductors connected to sensor pads. The capacitance changes as a function of the proximity of an object or objects to the sensor matrix. The change in capacitance of each node in both the X and Y directions of the matrix due to the approach of an object is converted to a set of voltages in the X and Y directions. These voltages are processed by digital circuitry to develop electrical signals representative of the centroid of the profile of the object, i.e, its position in the X and Y dimensions. Noise reduction and background level setting techniques inherently available in the architecture are employed.

Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone. The ILD is utilized by a noise reduction system to enhance the speech of the primary acoustic signal.

Disclosed is a system and method for providing an oscillating signal of relatively precise frequency without using a signal provided by a crystal as a reference. Disclosed is a feedback oscillator circuit configured to output an oscillating signal having a frequency defined by a reference signal. The oscillating signal can be sent to one or more circuits including at least one frequency sensitive element. The frequency sensitive element produces an output signal which depends on the frequency of the oscillating signal. A controller controls the reference signal in order to cause an attribute of the output signal to have a value within a desired range.

A video input processor is disclosed. The processor includes: an imaging signal-generating portion configured to image a subject and producing first imaging signals containing visible light components and a second imaging signal containing near-infrared light components; a gain adjustment portion configured to adjustably set a maximum gain value according to a relative magnitude between the visible light components and the near-infrared light components and adjust a gain for the first imaging signals at the set maximum gain value; and a noise reduction portion configured to reduce noises in the first imaging signals after the gain has been adjusted.

A noise reduction hood for attenuating the acoustic noise of cooling fans emanating from an electronic enclosure. The hood is comprised of a base that is constructed and arranged to be supported on the enclosure. The base has an air duct that is adapted to pass air therethrough in a first direction, and includes at least one noise reduction panel disposed in the air duct to attenuate acoustic noise emanating from the enclosure. In one embodiment, the noise reduction panel divides the air duct into a plurality of acoustical flow passages extending through the base. In another embodiment, the hood includes a cover, which may be detachable, that is adapted to divert the air from the first direction to a second direction. In a further embodiment, the hood includes a rim extending from the base for surrounding a portion of the enclosure to secure the hood to the enclosure. The hood may include a plurality of interlocking panels that are arranged in a grid pattern to form an array of acoustical flow passages.

Apparatus includes generator generating reference signal based on noise emitted from sound source, detector detecting level of reference signal and change in level, unit comparing change with threshold-value range and produce compared result, filter filtering reference signal, adaptive filter having variable filter coefficient, unit updating filter coefficient according to change of level of reference signal for obtaining an updated filter coefficient, unit stopping updating of filter coefficient in response to compared result when change falls outside threshold-value range, unit storing updated filter coefficient each time filter coefficient is updated, generator generating control signal using stored filter coefficient, unit generating control sound based on control signal, microphone detecting synthesis sound pressure of control sound and noise to produce an error signal, and unit setting stored filter coefficient to more accurate coefficient than stored filter coefficient based on error signal, and signal acquired by filtering control signal through filter.

Improved vocal signals are obtained in headsets and similar devices by including a microphone inside a chamber formed at least in part by the wearer's ear. This second microphone provides a reduced noise input signal. The reduced noise signal is corrected by input from another microphone, located outside the chamber. This correction can include echo cancellation, spectral shaping, frequency extension, and the like.

A method and system for enhancing a target sound signal from multiple sound signals is provided. An array of an arbitrary number of sound sensors positioned in an arbitrary configuration receives the sound signals from multiple disparate sources. The sound signals comprise the target sound signal from a target sound source, and ambient noise signals. A sound source localization unit, an adaptive beamforming unit, and a noise reduction unit are in operative communication with the array of sound sensors. The sound source localization unit estimates a spatial location of the target sound signal from the received sound signals. The adaptive beamforming unit performs adaptive beamforming by steering a directivity pattern of the array of sound sensors in a direction of the spatial location of the target sound signal, thereby enhancing the target sound signal and partially suppressing the ambient noise signals, which are further suppressed by the noise reduction unit.

An array of microphones utilizes two sets of two microphones for noise suppression. A primary microphone and secondary microphone of the three microphones may be positioned closely spaced to each other to provide acoustic signals used to achieve noise cancellation. A tertiary microphone may be spaced with respect to either the primary microphone or the secondary microphone in a spread-microphone configuration for deriving level cues from audio signals provided by tertiary and the primary or secondary microphone. Signals from two microphones may be used rather than three microphones. The level cues are expressed via an inter-microphone level difference (ILD) which is used to determine one or more cluster tracking control signals. The ILD based cluster tracking signals are used to control the adaptation of null-processing noise cancellation modules. A noise cancelled primary acoustic signal and ILD based cluster tracking control signals are used during post filtering to adaptively generate a mask to be applied against a speech estimate signal.

There is provided an image signal processing apparatus, comprising a demosaic processing unit receiving input of mosaic image data of each of signals obtained by a single plate imaging device having an element array composed of visible light obtaining elements obtaining visible light signals, and invisible light obtaining elements obtaining signals including invisible light components, and generating a demosaic image of each of the obtained signals; and a noise reduction processing unit receiving input of the demosaic image to execute correction of pixel values of the demosaic image obtained by the visible light obtaining elements on the basis of edge information extracted from the demosaic image of the signals obtained by the invisible light obtaining elements.

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with processor (30). Processor (30) receives aural signals from two sensors (22, 24) each at a different location. The two inputs to processor (30) are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source, and the spectral content of the desired signal are determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal with output device (90). Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

The invention discloses a pump potential cavitation fault detecting method based on quick spectrum kurtosis analysis. The pump potential cavitation fault detecting method comprises the steps that 1, vibration acceleration signals are collected to be subject to noise reduction to serve as to-be-processed signals; 2, according to the data size of the signals, the resolving order of signal processingis determined; 3, according to the quick spectrum kurtosis algorithm computation result, the optimal carrier frequency and bandwidth are selected; 4, the selected signals within the carrier frequencyand bandwidth are subject to Fourier transformation, and a frequency spectrum envelope diagram is obtained; and 5, an original signal time domain figure, a signal time domain figure treated through quick spectrum kurtosis filtering processing and the frequency spectrum envelope diagram obtained after selected zone Fourier transformation are compared, and cavitation trouble signal time and frequency characteristics are analyzed. By means of the method, more cavitation instant signals can be detected, information in the aspects of time domain and frequency domain is seen more clearly, and the normal state and the cavitation state of a pump can be distinguished obviously.

Noise reduction for application of structural test patterns to a Device Under Test (DUT) is accomplished with a capacitor “booster” bypass network on the probe card in which the capacitors are charged to a much higher voltage V_boost than the DUT power supply voltage VDD. Charging the capacitors to a voltage N×VDD allows the buster network to store N times the charge of a conventionally configured capacitance network, and effectively provides N times the capacitance of the original network in the same physical space.

A speech recognition system comprises: an input, for receiving an input signal from at least one microphone; a first buffer, for storing the input signal; a noise reduction block, for receiving the input signal and generating a noise reduced input signal; a speech recognition engine, for receiving either the input signal output from the first buffer or the noise reduced input signal from the noise reduction block; and a selection circuit for directing either the input signal output from the first buffer or the noise reduced input signal from the noise reduction block to the speech recognition engine.

A method comprises processing M subband communication signals and N target-cancelled signals in each subband with a set of beamformer coefficients to obtain an inverse target-cancelled covariance matrix of order N in each band; using a target absence signal to obtain an initial estimate of the noise power in a beamformer output signal averaged over recent frames with target absence in each subband; multiplying the initial noise estimate with a noise correction factor to obtain a refined estimate of the power of the beamformer output noise signal component in each subband; processing the refined estimate with the magnitude of the beamformer output to obtain a postfilter gain value in each subband; processing the beamformer output signal with the postfilter gain value to obtain a postfilter output signal in each subband; and processing the postfilter output subband signals to obtain an enhanced beamformed output signal.