36results 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.

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 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.

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.

A method for calibrating an acoustic antenna (10), the acoustic antenna comprising a plurality of acoustic transducers (11_k), each acoustic transducer being able to generate an electrical signal (s_k) under the effect of a detection of an acoustic wave, the antenna comprising elementary transducers (11_k) that are distributed over an antenna row or an antenna plane, about a reference transducer (11₀), the antenna defining a main axis (Y₀), passing through the reference transducer, and perpendicular to the antenna row or antenna plane, the method comprising the following steps:

• • a) placing a calibration source (5) in at least one position (r₀, r_j) with respect to the antenna, the calibration source being able to transmit a calibration acoustic wave (6);
  • b) measuring signals (s_k) generated by all or some of the elementary transducers in response to the calibration acoustic wave;
  • c) on the basis of the measurements performed in step b), determining a temporal phase shift (p⁰_k,n, p_k,j) of the signal respectively generated by each elementary transducer;
  • d) reiterating a) to c) in such a way that, in at least one iteration, the position of the calibration source may be considered to be centered on the main axis;
  • the method comprising estimating a phase shift (Φ_k) of each elementary transducer with respect to the reference transducer.

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.

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.