415results about How to "Improved noise suppression" patented technology

A system is disclosed that can determine the speed of an object and report that speed in a format suitable for use on a television broadcast, a radio broadcast, the Internet or another medium. One example of a suitable use for the present invention includes determining the speed that a baseball player swings a bat. Another use of the present invention is to measure the speed of a moving ball. In one embodiment, the system includes a set of radars positioned behind and pointed toward the batter. Data from all of the radars is collected and sent to a computer which can determine the start of a pitch, when a ball was hit, the speed of the ball and the speed of the bat.

A flow enhancement improvement for an underwater turbine generator (10) is disclosed wherein a longitudinal hole (240) is disposed in the central area (26), typically a hub (20) of the generator (10), and a second, augmentor duct (41), preferably rigid, is disposed about the outer duct (40) or housing of the unit to create a slot (200) area. The slot (200) and hollow hub (20) create areas of smooth, laminar fluid flow. The leading edges of the hub (20) or central ring and the augmentor (41) and outer ducts (40) are elliptical to enhance the fluid dynamics of the structure.

A bio-monitor is built into a telephone handset or cell phone. Sensors are configured to obtain bio-signals while the handset or cell phone is in the position for normal speaking use of the telephonic device. This enables biosignal acquisition and / or bio-signal telephonic transmission to occur without the need for a position change to effect voice communications. The invention can also be constructed in the form of a case or harness designed to fit over a preexisting cell phone or a pre-existing telephone handset.

The invention provides a method for enhancing microphone array voice based on combined inhibition. The method comprises the following steps of: structuring a microphone array for receiving external signals; analyzing the signals and obtaining time delays of different array signals relative to benchmark array signals in the microphone arrays opposite to a target voice source; respectively performing time delay compensation on digital signals corresponding to the two microphones, obtaining the compensated signals; respectively performing subband decomposition on the compensated array signals, and then forming fixed beams on each subband; meanwhile, respectively using blocking matrixes on each subband to obtain noise reference signals on each subband; and then respectively removing the noiseirrelative to the target voice form the fixed beam forming device on corresponding subband through an adaptive filtering processing algorithm, and then merging the subbands, thereby forming an initial gain signal; and meanwhile, making use of the previously compensated any two array signals to obtain a filter for inhibiting the noise signal related to the target voice through a recursive mutual power spectral density, thereby obtaining the final target voice signal through combining the initial gain signal.

The present invention is a method of processing seismic data in which one or more seismic vibrators are activated with one or more pilot signals and vibrator motions are recorded along with seismic data. Vibrator signatures are computed from measured vibrator motions, such as the ground force signal. A desired impulse response is specified from either a measured vibrator motion or from test data or field data from a location near the location from which the seismic data was acquired. A deconvolution filter is computed from the impulse response and the vibrator signature. Alternatively, a single separation and deconvolution filter is derived from the impulse response and from vibrator signatures from multiple vibrators and sweeps. The deconvolution or deconvolution and separation filter is used to process the seismic data. The vibrators are then moved to a new location, and the activation is repeated.

The present invention is a method of processing seismic data in which one or more seismic vibrators are activated with one or more pilot signals and vibrator motions are recorded along with seismic data. Vibrator signatures are computed from measured vibrator motions, such as the ground force signal. A desired impulse response is specified from either a measured vibrator motion or from test data or field data from a location near the location from which the seismic data was acquired. A deconvolution filter is computed from the impulse response and the vibrator signature. Alternatively, a single separation and deconvolution filter is derived from the impulse response and from vibrator signatures from multiple vibrators and sweeps. The deconvolution or deconvolution and separation filter is used to process the seismic data. The vibrators are then moved to a new location, and the activation is repeated.

A modular silencer includes an outer tube and a plurality of chamber separators suspended through a tensile force within a bore of the outer tube and defining chambers between each pair of chamber separators in the silencer. An outer edge of the chamber separators is spaced apart from an inner surface of the tube to allow gas equalization between the chambers. A plurality of tube portions separate the chamber separators from each other and from the proximal and distal ends of the silencer assembly. The tube portions define a continuous center tube suspended in tension within the outer tube and through which the projectile travels. The tube portions have can have angled openings through which gas discharge exits into the chambers. Rotatable sleeves can be disposed over the tube portions to further disrupt gas flow and dissipate heat and sound. A carbon fiber sleeve can be disposed over the outer tube to inhibit burn injuries from contacting the silencer.

The invention relates to a dual-microphone-based speech enhancement device. The device comprises a digital microphone array module, a first pulse density modulation decoder module, a directivity formation module, a second pulse density modulation decoder module, a speech enhancement processing module and an output module, wherein the digital microphone array module is used for receiving a time domain noisy speech digital signal; the first pulse density modulation decoder module is used for performing frame separation, down-sampling and filtering on the time domain noisy speech digital signal to form an initial down-sampling rate digital signal; the directivity formation module is used for performing directive formation on the initial down-sampling rate digital signal to form an initially noise-reduced directive output signal; the second pulse density modulation decoder module is used for performing the frame separation, the down-sampling, the filtering and accentuation on the initially noise-reduced directive output signal to form a second down-sampling rate digital signal; the speech enhancement processing module is used for performing speech enhancement processing on the second down-sampling rate digital signal; and the output module is used for outputting a background noise-removed speech signal. The invention also provides a dual-microphone speech enhancement device-based speech enhancement method. The device and the method have a good denoising effect.

Handheld apparatus (100), and method for comprehensive hearing testing with pass / refer results applicable for large scale neonatal screening, adult screening, full hearing diagnostic is disclosed. The apparatus (100) contains a signal processor (1), integral ear probe (150), and remote ear, and scalp probes (104) all packaged as a single handheld battery operated device (100). The apparatus (100) preferably performs a battery of tests, either independently or combined: oto-acoustic measurements utilizing a novel digital signal processing method for evoked oto-acoustic signal processing, auditory brain stem response test, tympanometry, and oto-reflectance. Algorithms for automatic test sequence, and pass / refer indication for the tests are provided.

The invention discloses a intra-pulse modulation and recognition method of a low signal-to-noise radar radiation source signal, which comprises the following steps of: receiving a radar radiation source pulse signal by an electronic reconnaissance receiver, after dimension reduction and A / D sampling process from radio frequency to intermediate frequency, obtaining a radar radiation source signal S(t) having different intra-pulse modulation modes, and processing the signal S(t) in a signal processing module, identifying and outputting intra-pulse modulation mode of the radar radiation source signal. With the method, the intra-pulse modulation mode of multiple radar radiation source signals can be correctly identified when the signal-to-noise is as low as -6dB; and compared with the conventional method for identifying multiple radar radiation source signals, the computational complex O (n2) or (n3) is lower.

The invention discloses a system and method for improving the out-of-band noise response of a multi-order delta-sigma modulator. The system and method includes programmable delta-sigma modulators which may be programmed to vary the reference signals at each modulator stage subsequent to the first stage relative to the reference signal of the first modulator stage. The resulting signal output will then typically exhibit the enhanced noise suppression characteristics of a dithered signal without the added circuitry and power required of a dithering apparatus.

Disposed at or toward the trailing edge of one or more nozzles associated with a jet engine are injection ports which can selectively be made to discharge a water stream into a nozzle flow stream for the purpose of increasing turbulence in somewhat of a similar fashion as mechanically disposed chevrons have done in the known art. Unlike mechanically disposed chevrons of the known art, the fluid flow may be secured thereby increasing the engine efficiency. Various flow patterns, water pressures, orifice designs or other factors can be made operative to provide desired performance characteristics.

A modular jack connector includes a housing defining a plug receiving space for receiving a complementary plug connector therein. The housing includes a printed circuit board with a number of first and second terminals mounted thereon. Each first terminal includes a spring arm extending into the plug receiving space. A first noise suppressing device is mounted on the printed circuit board and electrically connecting the first terminals with corresponding second terminals via the printed circuit board. This invention is characterized in that ferrite ceramics are integrally arranged to the second terminals so as to provide an enhanced noise suppressing result.

An integrated circuit memory is capable of storing analog information without the need for A / D conversion. Samples of a analog signal input are stored in nonvolatile memory cells. The integrated circuit is also capable of storing digital information in digital form. The sampling rate at which the analog signal input is sampled is user selectable. An internal signal path of the integrated circuit memory is differential, which enhances the precision with which the analog signal is stored in the memory cells.

A communication apparatus capable of echo cancellation is provided. The apparatus is configured to be provided with an input signal receiving interference. The input signal has a frequency range. The interference includes an echo component and a noise component. The apparatus has a first interference eliminator configured to reduce an echo component included in the input signal so as to produce an intermediate signal. The apparatus has a selector configured to select one of the input signal and the intermediate signal. The apparatus has a second interference eliminator configured to reduce at least one of a noise component and an echo component included in one of the input signal and the intermediate signal selected by the selector.

An apparatus for sound enhancement has at least two microphones (9) that provide a directional microphone array which is arranged to be pointed in the direction of a sound source. The directional microphone array thereby receives sound emitted by the sound source and generates sound signals. A processor (20) processors the sound signals generated by the microphone array to enhance the sound received by the directional microphone array from the sound source relative to other sound received by the directional microphone array. The processor (20) generates a corresponding enhanced signal (ES). Loud speakers (22) reproduce the enhanced signal as audible sound. Furthermore, sound suppression devices (7, 7a) are provided to suppress ambient should from reaching the eardrums of the user. This sound suppression acts in conjunction with the directional microphone array and the processor (20) which enhance the SOI to provide a listening environment in which the SOI is enhanced.

The present invention relates to a predistortion signal amplifier arrangement comprising digital predistortion means for predistortion of an input signal, a non-linear power amplifier, a predistortion control arrangement comprising error estimation means for estimating the error produced in the predistorted signal and a predistortion look-up table arrangement, the content in which is updated by the estimation means. The predistortion look-up table arrangement comprises at least two look-up tables, which are updated by said estimation means and said at least two look-up tables have different number of table entries. It further comprises a combiner for combining updated values obtained from said at least two look-up tables into a composite value and said composite value comprises a distortion coefficient which is input to said digital predistortion arrangement for signal predistortion.

An apparatus for sound enhancement has at least two microphones (9) that provide a directional microphone array which is arranged to be pointed in the direction of a sound source. The directional microphone array thereby receives sound emitted by the sound source and generates sound signals. A processor (20) processors the sound signals generated by the microphone array to enhance the sound received by the directional microphone array from the sound source relative to other sound received by the directional microphone array. The processor (20) generates a corresponding enhanced signal (ES). Loud speakers (22) reproduce the enhanced signal as audible sound. Furthermore, sound suppression devices (7, 7a) are provided to suppress ambient should from reaching the eardrums of the user. This sound suppression acts in conjunction with the directional microphone array and the processor (20) which enhance the SOI to provide a listening environment in which the SOI is enhanced.

The invention discloses a cooperated microphone voice control system and method of intelligent household. The control system comprises a signal collection module, a control center module and a cloud server, and the control center module further comprises a voice monitoring module, a data fusion module and a reverberation reducing preprocessing module. The control method comprises that the signal collection module collects voice signals; the control center module reduce noise of the voice signals; and the cloud server identifies the voice signals and analyzes a voice instruction. The voice signals are monitored, data is fused, and reverberation reducing preprocessing is carried out, so that the ambient noise and room reflection are effectively inhibited, voices are prevented from distortion to the largest degree, and the accuracy of subsequent voice identification can be ensured.

The invention relates to a deep learning noise reduction method fusing bone vibration sensor and microphone signals. The deep learning noise reduction method comprises the following steps: enabling abone vibration sensor and a microphone to collect audio signals, and respectively obtaining a bone vibration sensor audio signal and a microphone audio signal; inputting an audio signal of the bone vibration sensor into a high-pass filtering module, and carrying out high-pass filtering; inputting the bone vibration sensor audio signal subjected to high-pass filtering or the signal subjected to frequency band widening and a microphone audio signal into a deep neural network module; and obtaining the voice after noise reduction through prediction by the deep neural network module. According to the deep learning noise reduction method fusing bone vibration sensor and microphone signals, signals of a bone vibration sensor and a traditional microphone are combined, and the strong modeling capability of the deep neural network is utilized to realize high human voice restoration degree and extremely strong noise suppression capability, and the problem of human voice extraction in a complex noise scene can be solved, and the target human voice is extracted, and the interference noise is reduced, and a single-microphone structure can be adopted to reduce the cost. In addition, the signals obtained after the frequency band of the bone vibration sensor audio signals is broadened can be directly used as output.

The invention discloses a restraining device and restraining method for burst impulse noise of a power line communication channel. The restraining device comprises a Fourier transformation module, a masking subcarrier selection module and a compressed sensing estimation module. The restraining method for the burst impulse noise of the power line communication channel is based on compressed sensing and comprises the steps of carrying out Fourier transformation on a received signal to obtain a frequency domain signal, carrying out masking subcarrier selection to obtain a subcarrier only comprising background noise and the burst impulse noise, estimating the burst impulse noise in the power line communication channel by using a signal recover method based on compressed sensing, and finally, removing the impulse noise from the received signal to obtain a signal with noise removed. Due to the facts that the burst impulse noise in the power line communication channel has the characteristic of sparsity, and the OFDM masking subcarrier selection and compressed sensing technologies are applied, the effect of restraining the noise can be remarkably improved, and signal information of positions interfered by burst impulse can be effectively kept.

A method of antenna array processing. Digitized time-series data from a plurality of antenna elements are divided into a plurality of time windows. A Fourier Transform is performed on a time window of the plurality of time windows for each antenna element of the plurality of antenna elements to generate a plurality of frequency domain vectors. The plurality of antenna elements is divided into a plurality of largest aperture sub-arrays. The frequency domain vectors are compressive beamformed tor the each sub-array to generate a plurality of bearing estimates corresponding to the plurality of sub-arrays. The plurality of bearing estimates from the plurality of sub-arrays are sparsely combined to generate a plurality of frequency-specific bearing estimates. The plurality of frequency-specific bearing estimates are incoherently averaged over a range of frequencies to generate a bearing for a bearing-timing record. A plurality of bearings for the bearing-timing record is generated.

In a method for suppressing a noise by the spectrum subtraction method, it is possible to improve the noise suppression capability by simultaneously obtaining a frequency resolution required for the noise estimation spectrum and a temporal resolution required for the noise suppression spectrum. The signal length of an observation signal cut out for analyzing the spectrum of the observation signal used for estimation calculation of the noise spectrum is set longer than the signal length of an observation signal cut out for analyzing the spectrum of the observation signal as a value to be subtracted for performing subtraction with the noise spectrum.

An electromotive drive is provided, especially for the pump of a power-assisted steering system of a motor vehicle. The drive includes a housing (3) with a bearing journal (15) in which the shaft (18) of a rotor (9) is rotationally mounted. The drive also includes a stator (7) having drive windings. The bearing journal (15) extends through said stator and supports it. The stator (7) is supported by the bearing journal (15), substantially in the transversal direction only. The stator (7) is coupled with the remaining housing (3) in a rotationally fixed manner, so as to transmit the engine torque. The arrangement dramatically reduces disturbing noises which occur especially when the motor is operated at full load and which are caused by relatively high-frequency torque variations.

The invention provides an adaptive voice separating method based on sound source positioning, and relates to the technical field of information processing. The method includes steps: acquiring an audio signal of an observed environment, and confirming the number of sound sources and the direction of arrival of each sound source; generating a dimension reduction matrix P; generating a voice transfer matrix and a delay superposed wave beam coefficient; determining an active sound source of a frequency point and separating voice components; obtaining the obtained voice components and setting non-activated sound source components as zero; and obtaining time domain voice signals of the sound sources. According to the method, the number and the orientation of the sound sources in a current environment can be obtained through a sound source positioning technology, dimension reduction of each frequency band of the voice signal is performed with the cooperation of a PCA whitening technology toobtain an initial separation matrix, frequency components of each sound source channel are separated through the number of the activated sound sources at the frequency point by adaptive usage of the beam forming technology and the FDICA technology to restore the voice components, the obtained signal-to-noise ratio improvement characteristic is higher, better noise suppression performance is achieved, and the method is applicable to any sound source situations in the real voice environment.

The invention provides a speech noise suppression method and device, and relates to the technical field of speech signal processing. The speech noise suppression method firstly determines an acousticscene corresponding to a frequency domain speech signal according to a noise estimation result and a signal to noise ratio of the frequency domain speech signal, performs parameter adjustment on a noise processing model according to the acoustic scene, and then performs speech enhancement on the frequency domain speech signal according to the adjusted noise processing model. The speech noise suppression method adjusts the noise processing model based on a plurality of acoustic scenes to perform speech enhancement, so that the noise suppression has scene specificity, and the noise processing speed and the speech enhancement effect are improved.

The present invention provides a solid-state imaging apparatus that can significantly reduce kTC noise by using a negative feedback amplifying circuit. A solid-state imaging apparatus includes a pixel unit including a plurality of pixels arranged on a semiconductor substrate in a matrix, the pixel unit including, for each column, a source line and a column signal line, each of the plurality of pixels including: a photoelectric conversion unit that generates a signal charge corresponding to incident light; a storage unit storing the signal charge; a reset transistor; an amplifying transistor; and a cutoff transistor, wherein the amplifying transistor and the cutoff transistor form a negative feedback amplifying circuit. With this configuration, kTC noise can significantly be reduced.

Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for enhancing speech recognition accuracy. In one aspect, a method includes receiving geotagged audio signals that correspond to environmental audio recorded by multiple mobile devices in multiple geographic locations, receiving an audio signal that corresponds to an utterance recorded by a particular mobile device, determining a particular geographic location associated with the particular mobile device, generating a noise model for the particular geographic location using a subset of the geotagged audio signals, where noise compensation is performed on the audio signal that corresponds to the utterance using the noise model that has been generated for the particular geographic location.

The invention provides a magnetic resonance image reconstruction method. Compared with a prior art, the magnetic resonance image reconstruction method selects same or different convolution kernels to perform multiple iterative computations on imaging signal data so as to effectively remove noise in the signal data, achieve a more obvious noise suppression effect, and obtain images with higher signal-to-noise ratios. The invention further provides a magnetic resonance image reconstruction apparatus.