416 results about "Signal processing algorithms" patented technology

A fully digital camera system provides high-resolution still image and streaming video signals via a network to a centralized, server supported security and surveillance system. The digital camera for collects an image from one or more image transducers, compressing the image and sending the compressed digital image signal to a receiving station over a digital network. A plurality of image transducers or sensors may be included in a single camera unit, providing array imaging such as full 360 degree panoramic imaging, universal or spherical imaging and field imaging by stacking or arranging the sensors in an array. The multiple images are then compressed and merged at the camera in the desired format to permit transmission of the least amount of data to accomplish the desired image transmission. The camera also employs, or connects to, a variety of sensors other than the traditional image sensor. Sensors for fire, smoke, sound, glass breakage, motion, panic buttons, and the like, may be embedded in or connected to the camera. Data captured by these sensors may be digitized, compressed, and networked to detect notable conditions. An internal microphone and associated signal processing system may be equipped with suitable signal processing algorithms for the purpose of detecting suitable acoustic events and their location. In addition, the camera is equipped with a pair of externally accessible terminals where an external sensor may be connected. In addition, the camera may be equipped with a short-range receiver that may detect the activation of a wireless ‘panic button’ carried by facility personnel. This ‘panic button’ may employ infrared, radio frequency (RF), ultrasonic, or other suitable methods to activate the camera's receiver.

A fully digital camera system provides high-resolution still image and streaming video signals via a network to a centralized, server supported security and surveillance system. The digital camera for collects an image from one or more image transducers, compressing the image and sending the compressed digital image signal to a receiving station over a digital network. A plurality of image transducers or sensors may be included in a single camera unit, providing array imaging such as full 360 degree panoramic imaging, universal or spherical imaging and field imaging by stacking or arranging the sensors in an array. The multiple images are then compressed and merged at the camera in the desired format to permit transmission of the least amount of data to accomplish the desired image transmission. The camera also employs, or connects to, a variety of sensors other than the traditional image sensor. Sensors for fire, smoke, sound, glass breakage, motion, panic buttons, and the like, may be embedded in or connected to the camera. Data captured by these sensors may be digitized, compressed, and networked to detect notable conditions. An internal microphone and associated signal processing system may be equipped with suitable signal processing algorithms for the purpose of detecting suitable acoustic events and their location. In addition, the camera is equipped with a pair of externally accessible terminals where an external sensor may be connected. In addition, the camera may be equipped with a short-range receiver that may detect the activation of a wireless ‘panic button’ carried by facility personnel. This ‘panic button’ may employ infrared, radio frequency (RF), ultrasonic, or other suitable methods to activate the camera's receiver.

A sensing device and method of estimating the position and orientation of an object with respect to a local or a global coordinate system is disclosed. The method and device include one or more optical sensor, a signal processing circuitry and a signal processing algorithm to determine the position and orientation. A sensor is positioned within the housing. At least one of the optical sensors used in the method and system outputs information based at least in part on the detection of the signal of one or more light sources.

An omnidirectional sensor array system, for example a panoptic camera, comprising a plurality of sensors arranged on a support of predetermined shape to acquire data, wherein said sensors are directional and wherein each sensor is attached to a processing node which comprises integrated electronics that carries out at least a portion of the signal processing algorithms locally in order to reduce the computational load of a central hardware unit.

An ultrasound imaging system with pixel oriented processing is provided in which a method of producing a Doppler velocity image is accomplished by emitting unfocused acoustic signals into a medium over substantially an entire field; receiving scattered and reflected ultrasonic signals on a transducer array in response to the emission; processing the received ultrasonic signals to extract information to construct a Doppler velocity signal corresponding to at least one point in the medium; and generating on a display device the Doppler velocity image from the processed Doppler velocity signal. Acquisition sequences and signal processing algorithms are described that provide improved quantification of fluid flow parameters, including improved discrimination between regions of blood flow and tissue. Very high frame rate Spectral Doppler and Vector Doppler acquisition modes for real-time and post-acquisition visualization over a large field of view are described.

The invention relates to a system and a method for assessment of walking and miming gait in human. The method is preferably based on the fusion of a portable device featuring inertial sensors and several new dedicated signal processing algorithms: the detection of specific temporal events and parameters, 5 optimized fusion and de-drifted integration of inertial signals, automatic and online virtual alignment of sensors module, 3D foot kinematics estimation, a kinematic model for automatic online heel and toe position estimation, and finally the extraction of relevant and clinically meaning-full outcome parameters. Advantageously including at least one wireless inertial module attached to foot, the system provides common spatio-temporal parameters (gait cycle time, stride length, and stride velocity), with the 10 advantage of being able to work in unconstrained condition such as during turning or running. It furthermore may provide original parameters for each gait cycle, both temporal (load, foot-flat and push duration) and spatial (foot clearance and turning angle), and their inter-cycles variability. The system and method according to the invention allows the assessment of various aspects of gait which have shown recently to be of premium importance in research and clinical field, including foot clearance, 15 turns, gait initiation and termination, running, or gait variability. The system may be light weight, easy to wear and use, and suitable for any application requiring objective and quantitative evaluation of gait without heavy laboratory settings.

A gradiometer for measuring properties of a magnetic field and in particular, for measuring magnetic field gradient components, comprising at least two magnetic sensors wherein at least two of the magnetic sensors are arranged to sense the magnetic field component in substantially the same direction. The magnetic sensors may be super conducting quantum interference device (SQUID) magnetometers, Hall probes, flux gates or magneto-resistive magnetometers. The gradiometer also includes a computer processor loaded with an adaptive signal-processing algorithm, for performing adaptive signal balancing of the magnetometer outputs. In a preferred embodiment the gradiometer may comprise at least eight magnetometers in a three-dimensional arrangement, and a set of three orthogonal global feedback coils, one for each direction x, y, z, such that the five independent magnetic field gradient components may be measured. The gradiometer may also be used to measure second or higher order magnetic field gradient components.

An improved method for adaptively cancelling acoustic feedback in hearing aids and other audio amplification devices. Feedback cancellation is limited to a frequency band that encompasses all unstable frequencies. By limiting the bandwidth of the feedback cancellation signal, the distortion due to the adaptive filter is minimized and limited only to the unstable feedback regions. A relatively simple signal processing algorithm is used to produce highly effective results with minimal signal distortion.

The invention relates to a phased ultrasound locating method of the partial discharge of a transformer, which is used for monitoring and locating partial discharge faults. Phased array ultrasound sensors arranged at two different positions of the transformer are adopted to receive ultrasound broadband signals sent out by a partial discharge source, and an array model is formed; then a broadband focus algorithm in a broad-ban array signal processing algorithm is applied to convert broadband signals received by the two phased array ultrasound sensors into narrow-band signals, then a narrow-band direction finding algorithm is applied to respectively carry out direction of arrival (DOA) estimation on the two narrow-band signals, and finally a cross locating algorithm is applied to determine the concrete position of the partial discharge. Meanwhile, the invention also discloses a locating system. The invention can carry out on-line accurate locating on the partial discharge in the transformer; moreover, the used locating system has simple structure, and the method is simple and easy to apply.

A system and method for acoustic detection of coronary artery disease (CAD) are provided. The system includes a transducer for acoustically detecting heart signals of a patient and a computer system which executes detection software for processing the detected heart signals to identify the presence of CAD from the heart signals. The software allows for the automatic definition of a diastolic “window” of the acoustic signal for analysis, and automatically edits the sampled acoustic signal to eliminate unwanted artifacts and / or noise in the acoustic signal. The edited signal is then processed by a plurality of signal processing algorithms, including spectral analysis algorithms, time-frequency algorithms, global feature algorithms, kurtosis algorithms, mutual information algorithms, negenthropy algorithms, and principal component analysis algorithms, to generate a disease vector. The disease vector is then classified to determine whether CAD is present in the patient. Classification can be accomplished using linear discriminant analysis or a support vector machine.

The invention discloses a large-scale MIMO (Multiple Input Multiple Output) antenna array far field calibration system. The calibration accuracy of a wave beam direction error caused by an array element position error and an amplitude phase error in an antenna array under test can be effectively increased by the calibration system. The calibration system is implemented by the following technical scheme: a server control device controls the relative position between the antenna array under test and a beacon antenna, and changes a wave beam incident angle from a transmitting beacon antenna to areceiving antenna array under test or a wave beam departure angle from a transmitting antenna array under test to a receiving beacon antenna; a phased array wave control device controls all phase shifters, attenuators, switches and digital weighting of a feed network of the antenna array under test; a data transmitting and acquiring device controls digital signals transmitted by the transmitting antenna array under test and the transmitting beacon antenna, and acquires digital signals received by the receiving antenna array under test and the receiving beacon antenna; and the array element positioning error and the amplitude phase error of the antenna array under test are calculated on the basis of a digital domain signal processing algorithm, and a calibration result of the antenna arrayunder test is output.

The invention provides a smart grid-oriented power quality monitoring system and a smart grid-oriented power quality monitoring method, which relate to the field of monitoring of power quality. The system comprises a distributed online / offline node detection module, a network communication module, a distributed database module and a comprehensive management module of the power quality monitoring system, wherein the distributed online / offline node detection module comprises a parallel analog-to-digital conversion module and a signal processing module so as to realize distributed detection of asmart grid. In the technical scheme provided by the invention, a plurality of analog-to-digital conversion chips are connected in parallel and an analog signal is sampled, so that high-accuracy sampling rate of signals is increased and the detection requirement on transient power quality signals is met; based on a programmable logic chip, online analysis and data compression are realized by a parallel signal processing algorithm; a distributed database is adopted, so that system reliability is enhanced and system upgrade is promoted; and a global positioning system (GPS) and a communication network module are adopted, so that a distributed network system is realized, and power quality monitoring of the smart gird is supported powerfully.

The invention relates to a tinnitus hearing aid which comprises a tinnitus masking module and a hearing compensation module, and is characterized in that the tinnitus hearing aid further comprises a control module; the control module comprises a pre-setting unit, a selecting unit and a voice classification device; the pre-setting unit is used for selecting any one function mode in tinnitus masking, hearing compensation and tinnitus masking and hearing compensation, and inputting related preset parameters; the voice classification device is used for sequentially performing pre-treatment, feature extraction and classification according to received signals, and finally determining a sound field environment; a signal processing algorithm is selected and the parameters in the signal processingalgorithm are adjusted according to the function mode selected by the pre-setting unit and the determined sound field environment, and optimal combination is further performed; and the selecting unitis used for opening the voice classification device and working according to the function mode selected by the pre-setting unit.

A method of calibrating a robotic device, such as a cardiac catheter, includes oscillating the device on an actuation axis by applying an oscillation vector at an oscillation frequency. While oscillating, a location of the device is periodically measured to generate a plurality of location data points, which may express the location of the device relative to a plurality of measurement axes. The location data points are then processed using a signal processing algorithm, such as a Fourier transform algorithm, to derive a transfer function relating a position of the device to a movement vector for the actuation axis. The transfer function may be resolved into and expressed as a calibration vector for the actuation axis, which may include one or more components, including zero components, directed along each of the measurement axes. The process may be repeated for any actuation axes on which calibration is desired.

An electric blood pressure measurer without wound based on the algorithm for processing the pulse wave signals of inflatable cuff is disclosed. Its measuring steps include invoking the blood pressure measuring program module, running it, inflating the cuff to a certain pressure, releasing air by predefined steps, synchronously detecting the pressure of cuff and pulse wave for each step, monitoring and judging the time and pressure variation, determining if the measurement is stopped, processing the recorded pulse waves to obtain average pressure, and calculating systolic pressure and diastolic pressure.

The apparatus comprises: a signal modulating unit, a check meter circuit, a digital signal process unit, a display unit and a network interface unit. The analog output signals from the mutual inductor under test and the standard transformer are inputted to the signal modulating unit to make modulation; the modulated signals are inputted to the check meter circuit; the digital signals from the mutual inductor are directly inputted into the check meter circuit; the modulated analog signals and / or the un-modulated digital signals are processed by the check meter circuit to form the digital signals in clock and data synchronization; the obtained synchronous digital signals are outputted to the signal processing unit; after processed to get the amplitude, frequency and phase information of the tested signals, then figuring out the ratio difference and angle difference; comparing the index result obtained from the mutual inductor under test with the measurement result of the standard mutual inductor to evaluate the precise level of the mutual inductor under test.

An apparatus and method for time synchronization of a plurality of multimedia streams are described. In one embodiment, the method includes the concurrent capture of multimedia data via a plurality of multimedia platforms. During the concurrent capture of the multimedia data, each of the multimedia platforms receives a synchronization signal from a synchronization generation unit. In response, each platform processes a received synchronization signal to generate a common reference clock signal among each of the platforms. Once the common clock signal is generated, each of the platforms synchronizes captured multimedia data to form multimedia stream data according to the common reference clock signal. As such, the plurality of multimedia platforms are able to perform collaborative signal processing tasks of multimedia streams, including, for example, array signal processing algorithms.

A virtual multichannel sound system is presented to improve audio reproduction by statically or dynamically conforming signal processing to specific speaker characteristics and / or arrangements. According to one such aspect, one or more dynamic signal processing algorithms driving two or more speakers are altered in response to the relative physical characteristics or arrangements of these speakers, where parameter information for these algorithms is either factory set, user input, or automatically supplied to the processor. Examples of such relative speaker differences include speaker spacing or alignment, speaker or enclosure compliance, and enclosure configuration. Another aspect is to alter the processing algorithms in response to common speaker characteristics for certain conditions of input signals. An example of this aspect is to alter the signal processing to improve bass response as a function of bass content in the signals being presented to the speakers and speaker size as well as relative speaker position.

A system and a method are provided for estimating the average dielectric properties of a plurality of regions in space. The application of this technique is illustrated for determining the average properties of breast tissue. The knowledge of average properties is important when UWB microwave radar signal processing algorithms are used for tumor detection and localization. The method is an extension of a time-domain inverse scattering algorithm based on the finite-difference time-domain method. A hybrid conjugate gradient optimization is used to minimize a cost function defined between a measured and a calculated total electromagnetic field at a series of antennas. The output of the method is an average set of electromagnetic material parameters that describe specific regions of interest in either a non-dispersive heterogeneous medium or a dispersive heterogeneous medium.

The invention discloses an optimized virtual sound synthesis method based on headphone replay. The method includes the steps that according to the physical and geometric characteristics of a sound environment needing to be virtually created, the room impulse response (RIR) arriving at the position of a listener is determined; according to the arrival spatial orientation of direct sounds and reflected sounds in the RIR relative to the center of the head of the listener, the corresponding head-related impulse response (HRIR) and the direct sounds and the reflected sounds in the RIR are selected to be respectively subjected to convolution so that binaural room impulse response (BRIR) can be obtained; single-path signals of a virtual sound source and the BRIR are subjected to convolution so that binaural sound signals can be obtained; the binaural sound signals and the inverse function of a headphone transfer function HpTF are subjected to convolution so that headphone compensated binaural sound signals can be obtained and fed back to headphones for replay. With the method, a signal processing algorithm can be optimized, the calculation amount is reduced, the synthesized binaural sound signals are subjected to headphone compensation, the in-head positioning and sound image distorsion problems can be well solved, and the method is wide in application range.

Guidance to an operator to more accurately position electrodes upon a segmented heart model (SGM). The SGM is included in a map panel on a display screen. A catheter advanced into a beating heart supports one or more electrodes. During a single beat of the heart, an image is obtained with darkened portions corresponding to locations of the electrodes. The image is presented in the same map panel as the SGM. The current location of the electrodes is confirmed relative to the SGM, either manually or through automated software algorithms. EP data is captured that represents electrophysiological signals of the beating heart at the current location for each of the electrodes. A signal processing algorithm is applied to the captured EP data in view of the confirmed current location of the electrodes to result in a calculation that is mapped at the confirmed location of the electrodes.

The invention discloses a multi-grain parallel storage system and a storage (300) supporting array multi-grain parallel reading and writing. The storage system includes a reading and writing port (301) and the storage (300); the storage comprises W storage blocks (305) and a data strobe network (302); each storage block (305) is a two-dimensional array comprising a plurality of storage units (303); each storage line (304) in the two-dimensional array includes W storage units (303); each storage block can read and write one storage line (304) once; and w is 2 to the power of n, and n is a natural number. The storage system provided by the invention can support different data types of matrix array data parallel reading at the same time, thereby radically clearing the transpose operation requirement in the signal processing algorithm, and improving the executing efficiency of the signal processing algorithm.

The invention provides a digital beam forming and phase fitting method, and aims to provide a method capable of suppressing interference, improving the signal to interference and noise ratio and accurately controlling the phase of a beam forming signal. The method is implemented according to the technical scheme that a steering vector model obtains relative steering vectors by using online calibration, accurate absolute steering vectors are acquired by combining absolute phase responses, which are measured in a darkroom, of a reference array element, model parameters of the steering vector are fitted out by combining the existing absolute steering vectors, and an accurate spare steering vector of an antenna array in any direction is acquired; the second-order statistic of the array signal is iteratively calculated through an array signal statistic hardware calculation module, and sent to an array signal processing algorithm module together with an antenna array steering vector and a channel response to calculate an optimal beam weight; and finally, the array signal processing algorithm module imbeds the optimal beam weight into the digital beam forming module to perform real-time digital beam forming and phase fitting and process multi-channel array data and beam output in real time.

The invention provides an intelligent mattress system and a method for accurately acquiring human body vital sign data on a mattress. The system comprises an intelligent control device, an air pressure sensor and a mattress, wherein the mattress comprises an inflatable bag layer; and the intelligent control device is used for detecting and acquiring air pressure in the inflatable bag layer through the air pressure sensor, and acquiring the human body vital sign data on the mattress according to change of the air pressure. A user lies on the mattress, vibration caused by breathing and heartbeat results in corresponding change of the air pressure in the mattress, and change pressure signals are converted into change voltage signals through the air pressure sensor. Through a corresponding signal processing algorithm and a signal detecting algorithm, heartbeat and breathing frequency is extracted from the change voltage signals. The system is characterized in that the inflating amount of the inflatable bag layer of the mattress is adjusted to adjust softness of the mattress, and demands of different users to comfort level of the mattress are met; and breathing and heart rate, pressure and sleep quality of the user can be monitored through the intelligent control device module, so that the user can know his health data in time.

The present invention is in a “virtual RPM sensor” which is preferably implemented by means of a software process. The software determines the RPM of a rotating machine using an adaptive signal processing algorithm. Preferably, the input to the software system of the present invention is digitized waveform data from either a vibration transducer on a rotating machine or vibration data stored in a database or file. The present invention permits the determination of the RPM of a rotating machine without the requirement of additional hardware so that readings may be obtained inexpensively, accurately and without interrupting machine operation.

An endpoint processor includes a processor block, a timer block, a memory block, and analog-to-digital converter. The timer block is arranged to provide a time based signal to the processor block. The memory block cooperates with the processor block. The analog-to-digital converter is arranged to provide an interface between an analog signal and the processor block. The analog signal includes encoded data from a power signal. The processor block is arranged to control a sampling rate that is associated with the analog-to-digital converter such that the analog signal is down-converted as an under-sampled signal. The processor block is arranged to extract the encoded data from the down-converted signal by executing a digital signal processing algorithm that is stored in the memory block. The digital signal processing algorithm is arranged to reject fundamental and harmonic frequencies that are associated with a power-line frequency that is associated with the power signal.

The present invention relates to a binaural hearing aid system comprising a first hearing aid and a second hearing aid, each of which comprises a microphone and an A / D converter for provision of a digital input signal in response to sound signals received at the respective microphone in a sound environment, a processor that is adapted to process the digital input signals in accordance with a predetermined signal processing algorithm to generate a processed output signal, and a D / A converter and an output transducer for conversion of the respective processed sound signal to an acoustic output signal, and a binaural sound environment detector for binaural determination of the sound environment surrounding a user of the binaural hearing aid system based on at least one signal from the first hearing aid and at least one signal from the second hearing aid for provision of outputs for each of the first and second hearing aids for selection of the signal processing algorithm of each of the respective hearing aid processors so that the hearing aids of the binaural hearing aid system perform coordinated sound processing.

A method and an apparatus for detecting defects in the composite materials of a wind turbine blade or similar type of structure using an infrared (IR) camera. The temperature of the wind turbine blade is changed in such a way as to produce IR intensity changes in the region of the defect that can be visually detected or detected using a computer and signal processing algorithms. The same approach will work on other composite structures such as those found on aircraft.

A system and method for determining rotor speed of an AC induction machine is disclosed. The system is programmed to estimate a rotor speed of the induction machine according to a linear speed estimation algorithm and based on name plate information (NPI) of the induction machine and parameters of the AC induction machine during operation thereof. The rotor speed estimation system is also programmed to estimate a rotor speed of the AC induction machine according to a frequency-domain signal processing algorithm and determine if the rotor speed estimated thereby is valid. If the rotor speed estimated by the frequency-domain signal processing algorithm is valid, then a tuned rotor speed of the AC induction machine is estimated according to the linear speed estimation algorithm and based, in part, on the rotor speed estimated by the frequency-domain signal processing algorithm.