37results about "Systems with adjusting phase/time-lag-errors" patented technology

In one embodiment, an acoustic distance measurement system can use a modulation pattern to modulate a carrier wave. The modulated carrier wave may be transmitted to an acoustic transducer. An echo of the transmitted signal can be detected by correlating the received signal with the modulation pattern. Subsequent to detecting sufficient correlation between the received signal and modulation pattern, a distance determination can be made to determine the distance to the object that produced the received echo.

Disclosed herein is a multi-cantilever MEMS sensor functioning as a mechanical sensor having a plurality of cantilevers, replacing a conventional DSP based sound source localization algorithm and reducing production cost when the MEMES sensor applied to mass-produced robots, a manufacturing method thereof, a sound source localization apparatus using the multi-cantilever MEMS sensor and a sound source localization method using the sound source localization apparatus. The multi-cantilever MEMS sensor comprises a plurality of cantilevers 100 each of which includes a piezoresistor 20 and a sensing part 30 for sensing a predetermined signal generated according to the piezoresistor 20l; and a terminal T for detecting the signal generated according to the piezoresistor 20, wherein one end of each cantilever is a free end and the other end thereof is a fixed end of each cantilever, the piezoresistor 20 and the sensing part 30 are formed at the fixed end, and the free ends of the plurality of cantilevers 100 have different lengths. A method of manufacturing the multi-cantilever MEMS sensor is provided. Furthermore, a method of using the multi-cantilever MEMS sensor and a sound source localization apparatus are provided.

A computer system that processes mixtures of signals, such as speech and noise sources derived from multiple simultaneous microphone recordings, in order to separate them into their underlying sources. A source separation routine optimizes a filter structure by minimizing cross powers of multiple output channels while enforcing geometric constraints on the filter response. The geometric constraints enforce desired responses for given locations of the underlying sources, based on the assumption that the sources are localized in space.

The present invention is a highly accurate detector system for detecting and responding to wrong-way driving incidents. An optimum configuration and combination of sensors gather data corresponding to predetermined vehicle movement test parameters; a set number of parameter deviations from pre-determined thresholds will initiate various sensor signals. The sensor signals may be transmitted to the computer processor, which may in turn produce a range of system outputs. In various embodiments, system outputs may include but are not limited to outputs activating other system components, outputs initiating data storage and analysis, and outputs interfacing and communicating with other systems.

A system, processor and method of use for calibration processing is provided to calibrate acoustic vector sensor data collected at comparatively close range. Vector sensor data collected at close range includes data collected with source-to-receiver separations ranging from a one-tenth to approximately two acoustic wavelengths. The calculations substantially account for the acoustic impedance of a spherically diverging wave front, where the curvature is sufficiently pronounced to cause errors in resulting measurements in the calculations. The processing uses information contained within the vector sensor data to increase the accuracy of the vector sensor data.

The invention discloses an implementation method for a direction of arrival estimation system based on a smartphone sound array. The method comprises the following steps of firstly, deploying n mobile phones to a linear array at fixed intervals, establishing network communication and measuring a distance between the mobile phones; secondly, when a target sends a sound signal, simultaneously opening microphones of all n mobile phones and starting to record an audio signal of a target sound source; thirdly, closing the microphones after chirp signals which are played based on the mobile phones are recorded; fourthly, adopting a generalized cross correlation method to detect a time of arrival of the respective chirp signal in n segments of recorded audio signals; fifthly, synchronizing n segments of target sound source signals recorded by n mobile phones; and lastly, adopting an array signal processing method to carry out direction of arrival estimation on the audio signals of n segments of target sound source signals after synchronization in order to estimate an azimuth angle of the target. The method has the characteristics of easy and quick deployment, low cost and high precision of direction of arrival estimation and has wide application prospect in the fields of low-altitude unmanned aerial vehicle detection, field rescue and the like.

The invention provides a coprime matrix robust adaptive beamforming algorithm based on matrix filling. The coprime matrix robust adaptive beamforming algorithm based on matrix filling comprises the following steps that a sample covariance matrix of received data is calculated; the sample covariance matrix is vectorized to obtain vectors, and then elimination of redundancy and vector rearrangementare carried out on the vectors to obtain the received data vectors of a complete coprime matrix differential optimization matrix; zero is arranged between elements with discontinuous wave range difference in the received data vectors in a filling mode to obtain the vectors, and then the vectors are obtained by taking information of the positive half part of the vectors; the vectors are expanded into a Toplitz matrix; the Toplitz matrix is restored to obtain a filled covariance matrix; spectral peak search is carried out in an interference signal angle region to obtain estimation of arrival angles of each interference signal; the estimated arrival angles of interference signals, the physical array information of a coprime array and an interference and noise covariance matrix which reconstructs a coprime array physical array are utilized; weighted vectors of an adaptive beamformer is calculated by using the estimation of the interference and noise covariance matrix and expected signal steering vectors.

A computer system that processes mixtures of signals, such as speech and noise sources derived from multiple simultaneous microphone recordings, in order to separate them into their underlying sources. A source separation routine optimizes a filter structure by minimizing cross powers of multiple output channels while enforcing geometric constraints on the filter response. The geometric constraints enforce desired responses for given locations of the underlying sources, based on the assumption that the sources are localized in space.

The invention discloses an underwater direct-of-arrival estimation method and device based on a two-dimensional orthogonal non-uniform linear array. According to the method, phase compensation is carried out on a guide matrix of a non-uniform linear array to obtain a guide matrix of a uniform linear array corresponding to the non-uniform linear array; on the basis of an ESPRIT algorithm, DOA estimation iteration is carried out on the guide matrix of the uniform linear array to obtain rotation operators on uniform linear arrays corresponding to two non-uniform linear arrays; on the basis of a relationship between direction angles formed by acoustic waves reflected by a target on a non-uniform orthogonal linear array, an expression of a direct-of-arrival angle including the two rotation operators not including an acoustic speed is deduced; and then values of the two rotation operators are substituted into the expression to calculate the value of a final direction angle. According to theinvention, the propagation speed of the signal in the medium is eliminated during the calculation process, so that the influence on the algorithm result by the speed is eliminated. Measurement in a complicated underwater sound environment is realized; the practicability is improved; and the accuracy of the target positioning is high.

The invention provides a method and a system for infrasonic source orientation and speed control. The method comprises a step 101) of using a statistic analysis method, solving a effective delay range through a normal distribution model, that is, getting rid of all outliers in all delay values according to a mean value of a delay sum, so as to select an effective delay value; and a step 102) of based on the effective delay value, performing orientation and speed control processes on a infrasonic source by using a delay algorithm, to obtain orientation and speed control processing results of a to-be-tested infrasonic source by each sub-monitoring network. The method also comprises a step 103) of eliminating interference infrasonic sources based on the orientation processing results obtained by the sub-monitoring networks through a statistic comprehensive method, so as to complete optimized orientation on the to-be-tested target infrasonic source. The method is simple and reliable, and does not have harsh requirements on infrasonic monitoring formation and monitoring time, and meanwhile self noise of sensors and digital instruments can be eliminated, and interference generated by other infrasonic sources around the sub-networks can be eliminated.

The invention discloses a method for estimating a direction of arrival of a directional microphone microarray. The method comprises the steps of: (1) constructing the directional microphone microarrayto receive external sound wave analog signals, wherein the directional microphone microarray is composed of two microphones with directivity; (2) preprocessing the received analog signals by means ofthe two microphones with directivity respectively, and converting the analog signals into digital signals by means of an AD module; (3) performing noise reduction and voice enhancement processing onthe digital signals to obtain voice signals with a high signal-to-noise ratio; (4) comparing the acquired signal energy of the two microphones, judging the approximate direction of arrival by adoptingan energy comparison method, and partitioning the direction of arrival; and (5) solving voice time delay by using a generalized cross-correlation method, and calculating a precise direction of arrival by using a time delay estimation method. The method is simple in structure, reliable in performance, low in calculation complexity and suitable for a microphone solution under the condition that thenumber and the size of microphone arrays are greatly limited.

Systems and methods are described for determining a location of a user equipment (UE) within a wireless network using a reference signal of the wireless network. The disclosed systems and methods use multiple in-phase and quadrature (I / Q) samples generated from signals provided by receive channels associated with two or more antennas of the wireless system. The reference signal within the signal is identified from each of the receive channels based on the received reference signal parameters. An angle of arrival between a baseline of the two or more antennas and incident energy of the UE to the two or more antennas is determined based on the reference signals identified from each receive channel. The position of the UE is then calculated using the angle of arrival. The angle of arrival may be a horizontal angle of arrival and / or a vertical angle of arrival.

Systems and methods for determining user equipment (UE) locations within a wireless network using reference signals of the wireless network are described. The disclosed systems and methods utilize a plurality of in-phase and quadrature (I / Q) samples generated from signals provided by receive channels associated with two or more antennas of the wireless system. Based on received reference signal parameters the reference signal within the signals from each receive channel among the receive channels is identified. Based on the identified reference signal from each receive channel, an angle of arrival between a baseline of the two or more antennas and incident energy from the UE to the two or more antennas is determined. That angle of arrival is then used to calculate the location of the UE. The angle of arrival may be a horizontal angle of arrival and / or a vertical angle of arrival.

The invention relates to the field of direction-of-arrival positioning, and discloses a two-dimensional DOA estimation and channel phase disturbance correction method based on a uniform circular array. According to the two-dimensional DOA estimation and channel phase disturbance correction method, phase disturbance is processed at the cost of sacrificing the array aperture, and the method is a two-stage rank loss method. The two-dimensional DOA estimation and channel phase disturbance correction method comprises the steps of: 1, according to the characteristics of a Bessel function, truncatingan ideal steering vector in a compact uniform circular array mode space domain to construct a dimension reduction vector, and constructing a target function only related to phase disturbance based onthe dimension reduction vector and mode space domain conversion of channel phase disturbance; and 2, decoupling pitch and azimuth parameters in an ideal steering vector in a uniform circular array mode space domain to obtain a dimension reduction vector related to a pitch angle, and constructing a one-dimensional search target function only related to an azimuth angle. The two-dimensional DOA estimation and channel phase disturbance correction method provided by the invention can work online, and does not need an auxiliary array element. Besides, compared with a phase disturbance removing method based on a Hadamard product, the two-dimensional DOA estimation and channel phase disturbance correction method can be applied to a uniform circular array with a larger aperture, only needs a small number of array elements and can be applied to two-dimensional DOA estimation.

The invention provides an ultra-short base line orientation estimation method for any formation. The method comprises the steps of initializing a delay adjustment matrix according to a received signal and a delay signal thereof, thus acquiring a delay adjustment matrix function, and initializing a distributed search network; detecting the arrival time of the signals, determining synchronous moments, sampling the synchronous moments, building a synchronous sampling moment set, thus acquiring a reference array element sampling sequence of the signal arrival moments; and computing the rotated distributed search network, computing a matched peak of each direction vector under the synchronous sampling moment set according to the reference array element sampling sequence of the signal arrival moments and the delay adjustment matrix function, and traversing orientation angles, wherein the direction of the maximum matched peak is the final estimated orientation vector. The ultra-short base line orientation estimation method for any formation provided by the invention can be expanded to various complex formations, and is stable in performance when the part of array elements is invalid, and high in estimation accuracy of the distance and orientation of the target signal source.

The invention discloses an array calibration method based on time delay redundancy measurement, and the method comprises the steps: 1), carrying out the steps 2)-3) in sequence for a first auxiliary sound source S1 and a second auxiliary sound source S2: carrying out the band-pass filtering of a signal of one auxiliary sound source received by all array elements of a to-be-calibrated array A; 3) carrying out related time delay estimation on the combined signals of the two array elements of the array A to be calibrated after band-pass filtering processing to obtain a time delay vector relative to a reference array element; 4) establishing a position inversion objective function of each array element of the to-be-calibrated array A according to time delay vectors obtained for the first auxiliary sound source S1 and the second auxiliary sound source S2; and step 5) obtaining the coordinate of each array element through optimization search, thereby completing the calibration of the array A. According to the method, the array element space information is fully utilized, the array calibration precision and stability are improved, and the accuracy of the obtained sound source azimuth angle is higher when the method is adopted to calibrate the array detection target of the array.

The invention discloses a pool near-field double-hydrophone phase difference calibration and measurement system and a phase difference calibration and measurement method thereof, belongs to the technical field of underwater acoustic measurement, and aims to solve the problem that the position error has a great influence on the double-hydrophone phase difference calibration under a low-frequency near-field condition at present. The method comprises the following steps: rigidly mounting a low-frequency transmitting transducer in the middle of a bracket, rigidly mounting two hydrophones at the two ends of the bracket respectively, supplying a signal transmitted by a signal source to the low-frequency transmitting transducer through a power amplifier, and synchronously acquiring two paths of data through a dual-channel data acquisition unit; carrying out data processing on the acquired two paths of data to obtain the phase of each frequency point; solving the average value after multiple measurements, and acquiring the phase difference between the signals of two channels where the two hydrophones are positioned; and interchanging the two hydrophones, acquiring the phase difference between the signals of two channels where the two interchanged hydrophones are positioned, and solving the average value of the two phase differences to obtain the phase difference needing to be measuredby the measurement system. The phase difference calibration and measurement system and method are applied to phase difference measurement under a low-frequency near-field condition.

A beamforming system comprises an input module, a phase-difference module, a delay distribution module, and a delay estimation module configured to make a final delay estimation based on the delay distribution. The final delay estimation is applied to align one of two selected channels and combine the two channels to obtain a signal of interest. The phase-difference module calculates phase differences for a range of frequency bins. The delay estimation module considers all possible delays derived from the phase differences, including multiples of ±2π to overcome the ambiguity in high frequency bins when the spacing between two acoustic sensors is longer than the half of the wavelengths.

An acoustic system and method detect and locate low intensity and low frequency sound sources in an investigation area. An acoustic system is effective in identifying survivors trapped under rubble following a catastrophic event. The acoustic system focuses on the low frequency components of the human voice and includes acoustic sensors for detecting acoustic signals generated by the sound sources and for generating data representative of the acoustic signals. A wireless transmitter transmits the data representative of the detected acoustic signals to an electronic receiver block that receives and analyzes the data. A processor executes calibration of the acoustic sensors of the suite to temporally align each signal received from the acoustic sensors, and executes a digital beamforming to combine the data representative of the detected acoustic signals and to create an acoustic image of the investigation area to locate the low intensity and low frequency sound sources.

The invention discloses a target azimuth estimation method based on a low-frequency circular ring array, and belongs to the field of underwater acoustic signal detection. A hydrophone receives an underwater acoustic signal of 20Hz-5kHz; and an ADC converts an analog signal into a digital signal, performs spectral analysis, beamforming and azimuth estimation on the AD signal to obtain a signal target direction, receives an instruction of a serial port and outputs a beamforming result. Meanwhile, original data collected by the AD are stored in a large-capacity SD card through DDR. According to the method, the weight vector is stored in advance, so that the complexity of software is reduced, the operation period of the software is shortened, and the detection efficiency and real-time performance are improved; aiming at a coherent signal offset phenomenon, a method of injecting man-made noise, namely a generally diagonal loading technology, is adopted; by adopting a diagonal loading weighting coefficient, the beam forming robustness can be greatly improved under the condition of ensuring the high gain of the MVM method.

The present invention proposes a coprime array robust adaptive beamforming algorithm based on matrix filling, including: calculating the sample covariance matrix of the received data; vectorizing the sample covariance matrix to obtain a vector, and then performing de-redundancy and vector re- Arrange to get the received data vector of the complete coprime array differential optimization array; fill in all 0s between the discontinuous elements in the received data vector to obtain the vector, and then obtain the information of the positive half of the vector to obtain the vector; The vector is expanded into a Toeplitz matrix; the Toeplitz matrix is ​​restored to obtain a filled covariance matrix; the spectrum peak search is performed in the angle area of ​​the interference signal to obtain the estimation of the angle of arrival of each interference signal; the estimated angle of arrival of the interference signal is used, and The physical array information of the coprime array and the interference plus noise covariance matrix of the coprime array physical array are reconstructed; the weight vector of the adaptive beamformer is calculated by using the interference plus noise covariance matrix and the estimation of the desired signal steering vector.

There are provided examples of packages and methods for manufacturing the same. A package (100, 200, 400) as here described may comprise:- a package structure (102, 402) which defines inner surfaces (108a, 110a, 112a) delimiting an inner volume (104) and outer surfaces (108b, 110b, 112b) directed towards an exterior of the package (100, 200, 400);- at least one acoustic sensor element (120) applied to at least one of the inner surfaces (108a, 110a, 112a), to convert acoustic waves arriving from the exterior of the package (100, 200, 400) into acoustic information in the form of electric signals;- a plurality of millimeter wave sensing elements (140) applied to at least one of the outer surfaces (108b, 110b, 112b), to receive reflected radar signals from objects in the exterior of the package (100, 200, 400); and- a circuitry (160) applied to at least one of the inner surfaces (108a, 112a) of the package structure (102, 402), wherein the circuitry (160) is electrically connected to theat least one acoustic sensor element (120) and the plurality of millimeter wave sensing elements (140) to process the acoustic information and the reflected radar signals.