333 results about "Pulse response" patented technology

The invention is applied to the field of mechanical fault diagnosis signal processing, and particularly provides a pulse neural network mechanical fault diagnosis method based on STFT dimension transformation, and the method comprises the steps: collecting a one-dimensional mechanical vibration signal, carrying out the wavelet decomposition, carrying out the wavelet reconstruction of a low-frequency component and a denoised high-frequency component, and carrying out the wavelet reconstruction of the low-frequency component and the denoised high-frequency component; obtaining a denoised one-dimensional vibration signal; performing short-time Fourier transform, and converting the time-frequency two-dimensional matrix into a time-frequency two-dimensional matrix; inputting the time-frequency two-dimensional matrix into an improved HH threshold neuron model, carrying out Poisson sparse coding on the time-frequency two-dimensional matrix, and only carrying out pulse response on signal significant features; constructing a suprathreshold coding convolutional network with residual connection, inputting a sparse coding matrix, training by adopting an unsupervised learning rule based on STDP, and adaptively adjusting a network synaptic weight; and inputting to a trained above-threshold coding convolutional network, and obtaining pulse emission activity of neurons of an output layer through network forward propagation to determine a fault diagnosis result.

The invention discloses an indoor UWB-IMU fusion trajectory tracking system and method based on factor graph optimization, and belongs to the technical field of UWB indoor positioning. The system adopts a multi-stage processing framework, and specifically comprises the following steps: firstly, collecting channel pulse response and distance information by virtue of a wireless communication system, constructing a dynamic environment sensing mechanism, and evaluating the current channel state and the availability of each base station in real time; secondly, for the ranging values of the available base stations in the NLOS environment, compensating by using historical information, and for the ranging values of the unavailable base stations, estimating and correcting by using a time step length; and finally, collaborative optimization is carried out on UWB ranging data, IMU inertial information and base station state information through an SW-FGO algorithm, so that positioning stability is ensured, and adverse effects of obstacle shielding and a multipath effect on trajectory tracking precision are reduced. The invention provides a fusion positioning system which combines the advantages of UWB and IMU, can realize the tracking of a target motion track, and is suitable for most LOS and NLOS indoor mixed scenes.

A method in a network node and a network is provided to determine line of sight, LOS, base stations for a user equipment (UE) is provided. A request is provided to at least one of the UE and a plurality of base stations to measure and report LOS detection measurements, wherein the plurality of base stations includes base stations of a serving cell of the UE. The LOS detection measurements are received from the at least one of the UE and the plurality of base stations. LOS base stations for the UE are determined based on the LOS detection measurements. An indication of the LOS base stations is transmitted to the UE.

The invention discloses a distribution network power line anti-interference communication method based on adaptive filtering, and belongs to the technical field of distribution network power line communication. According to the method, a communication signal adopting a frequency shift keying modulation mode is acquired from a power line coupler of a distribution network through a signal acquisition module, the communication signal is preprocessed by using a digital signal processing unit, spectral analysis is performed on the preprocessed signal, and characteristic parameters of interference noise are extracted; a coefficient vector of a finite impulse response adaptive filter is initialized based on the characteristic parameters, the order of the filter is in direct proportion to the reciprocal of the dominant interference frequency, fine adjustment is carried out based on noise energy distribution through a table look-up method or a linear mapping relation, and the weight coefficient of the filter is adjusted in real time in the communication process. The adjustment step length is dynamically adjusted according to the real-time estimated signal-to-noise ratio, and finally, the processed signal is output through the adaptive filter and data demodulation is carried out. The method is mainly used for improving the anti-interference capability and the communication reliability of the power private line communication of the distribution network.

The invention discloses a GNSS element signal interference suppression method and system based on double complex number modeling, and the method comprises the steps: firstly introducing a double complex number model, carrying out the modeling of a received multi-frequency multi-path signal, and obtaining a GNSS element signal; secondly, carrying out modular operation projection on the GNSS element signal to obtain time-frequency domain analysis of the element signal, and obtaining a ridge value through detection; and correcting the ridge value to obtain instantaneous frequency estimation and instantaneous frequency modulation slope estimation of the element signal interference carrier and subcarrier, and applying the instantaneous frequency estimation and the instantaneous frequency modulation slope estimation to a finite impulse response filter to obtain the finite impulse response filter based on double complex numbers. And finally, performing double complex Kalman filtering on the received element signal to obtain an interference estimation value, removing the interference estimation value from the received element signal, and taking a real part to obtain a GNSS expected signal. According to the method, the high-precision interference is estimated by using the high-precision interference parameter estimation value and the double-complex Kalman filtering, and interference suppression in a multi-frequency multi-path environment is efficiently realized.

Systems, methods, software, and devices are disclosed herein that transform spatial input into modal output comprising learned modal components of an impulse response. A neural network interpolates the modal components of the impulse response based on a desired sound source direction represented in the spatial input. The learned modal components are then used to determine coefficients for an infinite impulse response filter that transforms anechoic audio into spatialized audio. The spatialized audio provides a directional effect to a listener as having arrived from the desired sound source direction.

The invention belongs to the technical field of artificial intelligence, and discloses an energy data acquisition encryption and privacy protection method and system. The power grid frequency deviation FREQ of a power line carrier communication system is obtained at the time of power grid data acquisition, and deviation information at different time points is combined into a random number sequence by using a hash function; a specific pulse signal is sent at a data acquisition point to obtain PLC channel pulse response CIR, a minimum mean square error estimation algorithm is used for processing, and key features such as the maximum pulse amplitude are extracted to serve as channel feature values. And then, inputting the random number sequence and the channel characteristic value into a deep reinforcement learning model trained by a large amount of historical data to generate an encryption key. And finally, encrypting the acquired energy data by adopting an AES algorithm based on the key, transmitting the energy data through an OFDM technology, and meanwhile, generating the key again according to the change of a channel characteristic value by setting a self-adaptive security mechanism. According to the method, the safety and privacy of energy data acquisition and transmission are remarkably improved.

This disclosure relates to techniques for performing beam selection using machine learning assistance in a wireless communication system. A wireless device and a cellular base station may establish a wireless link. An artificial intelligence model to use for beam selection may be determined. Channel impulse response information for a cell in a first frequency range may be determined. A downlink beam for a cell in a second frequency range may be selected based at least in part on the artificial intelligence model and the channel impulse response information for the cell in the first frequency range.

Serializer / deserializer (SerDes) modules, equalizers, and equalization techniques having robust parameter initialization may substantially reduce convergence time. One illustrative equalizer includes: a discrete-time, finite impulse response (“FIR”) filter to convert a receive signal to a filtered signal; a decision element to determine channel symbols represented by the filtered signal; and a controller. The controller is configured to, for each of multiple values in a search range, estimate a performance of the equalizer based on the channel symbols and at least one of the filtered signal or an input signal to the decision element; configured to find a centroid value based on the performance for each of the multiple parameter values in the search range; and configured to derive an initial value for the parameter from the centroid value.

This disclosure provides methods, components, devices and systems for adjacent channel interference (ACI) mitigation techniques for ultra-wideband (UWB) systems. Some aspects more specifically relate to ACI mitigation for UWB systems. In some implementations, a wireless communication device may receive control information that identifies a preamble for communications to the wireless communication device in a UWB system. The wireless communication device may receive, via a signal in the UWB system, a data packet that includes the preamble. The wireless communication device may estimate a channel impulse response (CIR) of the signal and determine an interference metric that is based on the estimated CIR. The wireless communication device may process the data packet based on the interference metric.

The invention relates to a signal behavior model conversion method and device, computer equipment, a computer readable storage medium and a computer program product. The method comprises the following steps: acquiring an initial scattering parameter in a frequency domain parameter model, and establishing an equivalent passive circuit model according to the initial scattering parameter; wherein the initial scattering parameter refers to a frequency domain scattering matrix parameter used for describing a signal transmission path; determining direct current point information according to the equivalent passive circuit model, and correcting the initial scattering parameter according to the direct current point information to obtain a corrected scattering parameter; determining corresponding pulse response data according to the corrected scattering parameter, and updating the corrected scattering parameter according to the pulse response data to obtain a target scattering parameter; and generating a target circuit model according to the target scattering parameter. By adopting the method, the conversion precision of the S-parameter-based circuit model can be improved.

Aspects of the present disclosure include a biometric device configured to receive biometric signals from sensors placed on a subject, and attenuate at frequencies causing noise in the biometric signals. Aspects are directed to a finite impulse response (FIR) filter configured to attenuate noise frequencies caused by line frequencies and / or flicker frequencies. Aspects of the FIR filter may be configured based on an analog-to-digital (ADC) sampling rate.

The embodiment of the invention provides a control method for sound partitions in a space, electronic equipment, a vehicle and a medium, and the method comprises the steps: determining a first sound source pulse response of a target sound field, the first sound source pulse response being a pulse response of a preset sound source in front of the target sound field; and based on an electro-acoustic transfer function of a bright area, the first sound source pulse response, an electro-acoustic transfer function of a dark area and an optimization function, determining a driving signal of each loudspeaker, namely, restraining the first sound source pulse response as a pulse response of a preset sound source in front of the target sound field, and finally, determining a more accurate loudspeaker driving signal through an optimization function. Therefore, it is clear that the sound image position of the replay sound field is right ahead of the listener, and the listening comfort is improved.

The invention relates to a method for operating a transceiver device (1), comprising the following steps to be performed by the transceiver device (1): transmitting a pulse radio signal (5) at a given time; receiving a reflected signal of the transmitted pulse radio signal (5); generating a channel impulse response that describes the reflected signal as a function of a path delay; determining a self-interference distance range of the path delay of the channel impulse response, wherein the reflected signal of the self-interference distance range describes a self-interference signal of the transceiver device (1); determining at least one target distance range of the path delay of the channel impulse response, wherein the reflected signal of the target distance range describes a signal of the target distance range;Determining a signal component of the target distance range signal correlated with the self-interference signal using an operator, wherein the operator maps the self-interference signal to the correlated signal component of the target distance range signal; determining a signal component of the signal uncorrelated with the self-interference signal by removing the signal component correlated with the self-interference signal using a correlation cancellation filter.

The invention relates to the technical field of sound correction, and discloses a room sound calibration method based on listening tendency control and related equipment, which are used for improving the matching degree of a calibration result and listening preference of a user. The room sound calibration method based on listening tendency control comprises the following steps: obtaining a listening tendency parameter selected by a user through an interactive interface and a room pulse response of a current room; determining a group of frequency-related window control parameters according to the hearing tendency parameter and the room pulse response; determining a windowing strategy with the window length changing along with the frequency based on the window control parameters; windowing the room impulse response by using a windowing strategy to extract an effective frequency response curve for acoustic calibration; calculating according to the difference between the effective frequency response curve and the target frequency response curve to obtain a correction filter coefficient; and calibrating the audio signal of the audio playing system based on the correction filter coefficient.

The present invention provides a method for low-complex estimation of Direction of Arrival (DoA) and Time of Arrival (ToA) in multi-antenna-based communication comprising detecting overlapped paths between a Mobile Station (MS) and a Base Station (BS) operating under said multi-antenna-based communication with scatters present therein involving channel paths represented as Channel Impulse Response (CIR) defined with channel signatures including delay and angle steering vectors, decoupling the channel paths in delay and angle-domain and separately estimate the DoA and ToA of each path with a low computational complexity which is based on one-dimensional low-index based rotation methodology and finally pairing the DoA and the ToA.

The invention discloses a sound velocity measurement method and device, a storage medium and electronic equipment. In the sound velocity measurement method provided by the specification, each array element in an ultrasonic transducer is utilized to sequentially emit an original emission signal, and aiming at each emission, all array elements in the ultrasonic transducer are utilized to receive a signal after the original emission signal penetrates through a to-be-measured tissue, so that an original measurement signal is obtained; aiming at each array element of the ultrasonic transducer, determining the pulse response of the array element according to the original transmitting signal and the original measuring signal; according to a preset extrapolation emission signal and each pulse response, determining a corresponding extrapolation measurement signal when each array element simulates the extrapolation emission signal; and determining the sound velocity in the tissue to be measured according to the original emission signal, the original measurement signal, the extrapolation emission signal and the extrapolation measurement signal.

The invention relates to the technical field of audio sound calibration, and discloses a room sound calibration method based on intermediate frequency reverberation time and related equipment, which are used for improving the accuracy of a sound calibration result. The room sound calibration method based on the intermediate frequency reverberation time comprises the following steps: acquiring a room impulse response of a target room and calculating the intermediate frequency reverberation time; according to the intermediate frequency reverberation time, the recommended window length of the frequency correlation window is determined by querying a predefined mapping relation, and the mapping relation is established based on acoustic association between the room reverberation time and the analysis window length; based on the recommended window length, determining the actual window length of the actual application in combination with the user setting mode; performing time domain windowing and frequency-related transfer function estimation processing on the room impulse response according to the actual window length to obtain window function weighted transfer function estimation; filter calibration coefficients are generated based on the transfer function estimates and applied for audio calibration.

A system for beamforming and filtering. In some embodiments, the system includes an array of antenna elements; and a front-end circuit connected to the antenna elements, the front-end circuit including a plurality of delay elements. The delay elements may be configured to provide beam forming; and to operate as delay elements in a finite impulse response filter.

The invention relates to the technical field of wireless ranging, and discloses a channel estimation method, a coherent combination method, a circuit, a sensor and equipment. The channel estimation method for ultra-wideband ranging comprises the steps that channel estimation is carried out on at least two fragments, channel impulse responses of the at least two fragments are obtained, and the at least two fragments are from the same MMS data packet; and accumulating the channel pulse responses of the at least two segments to obtain the channel pulse response of the MMS data packet. The channel estimation of the ranging packet based on the plurality of segments is realized, the energy of the channel pulse response is increased, the ranging sensitivity is improved, and the advantages of the ranging packet based on the plurality of segments are brought into full play.

The invention discloses a self-supervised ultra-wideband positioning method based on an improved double-delay depth deterministic strategy. The method comprises the following steps: acquiring channel impulse response data CIR and performing preprocessing; an improved double-delay depth deterministic strategy model is constructed to carry out UWB ranging error correction, the improved double-delay depth deterministic strategy model is based on a double-delay depth deterministic strategy gradient model, a double-commentator network structure is designed, a spatial-temporal feature coding layer based on a self-attention mechanism is introduced, and UWB ranging error correction is carried out. Capturing a long-range dependency relationship through a dynamic weight distribution mechanism, and establishing cross-timestamp time-space association feature mapping; a target network delay updating mechanism is introduced, so that the updating frequency of a target network is lower than that of a commentator network and an actor network, and action noise generated by a strategy is regularized; and generating a pseudo tag by adopting extended Kalman filtering, and carrying out iterative optimization to obtain a positioning result. The method is particularly suitable for processing noise and multipath effect in CIR data, and high-precision UWB positioning is realized.

Backscatter AMP device detection and characterization using Ultra Wideband (UWB) impulse radar may be provided. First, a computing device may transmit an UWB signal comprising pulses to a Backscatter Device (BKD). Next, the computing device may receive a reflection of the UWB signal from the BKD in response to the UWB signal. Then the computing device may locate the BKD using a Channel Impulse Response (CIR) of the reflection of the UWB signal.

Techniques are provide for neural network based positioning of a mobile device. An example method for measuring a channel in a wireless communication system includes: receiving reference signal information; determining one or more channel frequency responses based on the reference signal information and one or more timing hypotheses; determining one or more channel impulse responses comprising a channel impulse response for each of the one or more channel frequency responses; processing the one or more channel impulse responses with a neural network; and determining an output of the neural network.

The invention provides a far-field voiceprint recognition method and device, electronic equipment and a storage medium, and the method comprises the steps: collecting sample near-field audio data, constructing a geometric acoustic propagation model containing a frequency band related reverberation model and a sound source and receiving equipment directionality model based on the data, and generating a room pulse response function; performing convolution on sample near-field voice and the pulse response function to generate simulated far-field audio data; carrying out pre-training based on the sample near-field audio and carrying out fine tuning based on the simulated far-field audio by adopting a speaker representation weight pre-retention training strategy, and constructing a far-field voiceprint recognition model; and inputting the to-be-detected far-field audio after reverberation removal and the reference near-field audio into the model, calculating the similarity and outputting an identification result. The problems of scarcity of far-field voice data, poor model cross-domain adaptability, serious signal reverberation and the like are solved, and the accuracy and robustness of a far-field voiceprint recognition system are remarkably improved.

A UWB tag (10) for detecting the presence of a living object (20) is described, wherein the UWB tag (10) comprises at least one UWB front-end transceiver (11) and is configured to transmit multiple pulses (30) and receive multiple associated reflected channel pulse responses (32), and subsequently evaluate them for changes in the received associated reflected channel pulse responses (32), wherein the detection of the change in the associated reflected channel pulse responses (32) indicates the presence of the living object (20). Furthermore, a method (100) for detecting the presence of a living object (20) using at least one UWB tag and a multi-statistical radar system are described.

It is proposed to process sound signals acquired by an array of microphones and generated by a sound source to acoustically characterize a space (ESP) comprising the array (MIC) and the sound source and delimited by walls (PAR). A time-frequency transform is applied to the acquired signal and a generalized velocity vector is expressed in the frequency domain based on the acquired signal. Specifically, the generalized velocity vector is represented in the time domain v (t) in the form of a series of peaks comprising at least one peak associated with reflections from the wall and time abscissas associated with the delay TAU1, and the expression of the generalized velocity vector in the time domain is modeled by autoregressive moving average ARMA, the autoregressive moving average (ARMA) is defined by an autoregressive filter (AR) and a moving average (MA). Thus, it is provided that the acquired sound signal is processed by applying an autoregressive filter (AR) to obtain an impulse response characterizing the space (ESP) and obtained from the moving average (MA).

The application discloses a wireless communication channel parameter measurement method and system, comprising the following steps: obtaining a sending signal based on a PN sequence of a wireless transmitting end; obtaining an influence of a channel impulse response based on the sending signal; performing convolution on the sending signal and the influence of the channel impulse response to obtain a receiving signal; obtaining autocorrelation outputs of different channels based on the receiving signal and the PN sequence; and obtaining path loss of the channel based on the autocorrelation outputs of the different channels. Compared with the traditional channel measurement activity using a special channel measurement instrument or a VNA, the device cost is greatly reduced, the manpower and material resources cost of the measurement activity is low, and a large amount of channel data can be obtained according to different experimental schemes.

The invention provides a method and a device for realizing an audio delay effect based on an impulse response sequence and application. The method comprises the following steps: inputting an impulse response of audio data with an expected delay effect; the pulse response with the expected delay effect is preprocessed, N discrete pulse responses and the initial delay time corresponding to each discrete pulse response are extracted, and the main period of a pulse response sequence is determined; respectively carrying out convolution on an input audio signal and each discrete pulse response to generate a convolution signal corresponding to each discrete pulse response; calculating the actual delay amount corresponding to each discrete pulse response based on the initial delay time corresponding to each discrete pulse response, the main period of the pulse response sequence and the set adjustable target delay time; the convolution signal corresponding to each discrete pulse response is delayed by the corresponding actual delay amount, and N paths of signals are output; and mixing the output N paths of signals to generate a final delay effect output signal, and obtaining a high-quality delay effect in real time.

One or more measurements of a positioning reference signal (PRS) is performed (1010, figure 10) by a first apparatus 202 (operating as a terminal device, UE 110, or a base station, gNB 120). The first apparatus obtains (1020), based on the one or more measurements, at least one of: a set of first estimates 215 of a power delay profile (PDP), or a set of second estimates of a channel impulse response (CIR). The first apparatus performs (1030) differential quantization 220 for at least one of: a subset of first estimates from the set of first estimates, or a subset of second estimates from the set of second estimates. Then, the first apparatus transmits (1040), via a plurality of reference signal received power per path (RSRPP) messages, to a second apparatus 204 (operating as a location server, LMF 130), a measurement report 225 including at least one of: quantized first estimates of the PDP or quantized second estimates of the CIR. The UE may measure received DL PRS signals, or the gNB may measure received UL PRS signals, and estimate PDP or CIR. The container to encapsulate the CIR / PDP is not limited to RSRPP, and may be other reporting messages. The second apparatus 204 may perform 235 differential de-quantization for the at least one of the quantized first estimates or the quantized second estimates, and then construct 240 at least one of the power delay profile or the channel impulse response based on at least the differential dequantization.

This system for voice data processing comprises: an analog voice processing unit that receives an input of an analog voice signal and generates a digital voice signal; and a feature extraction unit that, when the digital voice signal is input, operates as an infinite impulse response (IIR) filter to filter the digital voice signal, and operate as an RNN to extract a voice feature from the filtered digital voice signal.