234 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.

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.

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 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 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.

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.

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.

A continuous time linear equalizer (CTLE) includes a first circuit path and a second circuit path. The first circuit path has a first step response. The second circuit path is in parallel with the first circuit path. The second circuit path has a second step response with a pulse response, and includes a transmission line that is configured to control a pulse width of the pulse response according to a length of the transmission line. An output signal of the CTLE is derived from an output signal of the first circuit path and an output signal of the second circuit path.

Systems and techniques for localization of disturbances are illustrated. One embodiment includes a method for detecting a disturbance in an area of operation, the area of operation including a plurality of transmitter-receiver pairs. The method obtains, based on a first plurality of signal transmissions between the plurality of transmitter-receiver pairs, at least one baseline channel impulse response (CIR) measurement. The method obtains, based on a second plurality of signal transmissions between the plurality of transmitter-receiver pairs, at least one additional CIR measurement; The method obtains, based on the at least one additional CIR measurement and the at least one baseline CIR measurement, at least one residual CIR measurement. The method derives, based on a sufficient similarity between the at least one residual CIR measurement and at least one particular key of a plurality of keys, localization data for at least one disturbance detected within the area of operation.

Various aspects of the present disclosure generally relate to wireless communication. In some aspects, a user equipment (UE) may receive a first indication of a set of analog receive beams and an instruction to report one or more post-analog-beamformed channel impulse responses (CIRs) that are based at least in part on the set of analog receive beams. The UE may transmit a second indication of a CIR report that indicates the one or more post-analog-beamformed CIRs. Numerous other aspects are described.

Aspects presented herein may improve the efficiency and performance of artificial intelligence (AI) / machine learning (ML) (AI / ML) positioning by enabling a network node (e.g., a base station, transmit receive point (TRP), etc.) to compress uplink (UL) reference signal measurements to reduce reporting overhead for UL reference signal measurements. In one aspect, a network node performs at least one channel impulse response (CIR) measurement or at least one channel frequency response (CFR) measurement for a sounding reference signal (SRS) set. The network node compresses at least one CIR measurement or at least one CFR measurement for the SRS set. The network node reports, for the network entity, one or more of the at least one compressed CIR measurement or the at least one compressed CFR measurement for the SRS set.

Anti-interference device for satellite radionavigation signal receiver The invention relates to an anti-interference device (10, 30, 40, 50) comprising: - an analog-to-digital converter (12) configured to transform a received analog radionavigation signal into a digital signal comprising a plurality of samples;- at least one pre-cumulative filter (22) adapted to a modulation of the transmitted signal, each filter corresponding to an impulse response filter, substantially equal to a waveform of the transmitted signal, allowing for the accumulation of samples and operating at the sampling frequency of the received signal, and delivering at the output a filtered complex signal, said device further comprising at least one first amplitude processing module (24) configured to process the amplitude of said filtered complex signal, said first amplitude processing module being both downstream of said pre-cumulative filter and upstream of at least one signal demodulation channel. Figure for the abbreviation: Figure 1;

This optical fiber wireless transmission system comprises: an optical transmitter (10) that converts a wireless signal and a monitoring swept-frequency signal in a frequency band different from the frequency band of the wireless signal into an optical signal having an intensity and outputs the optical signal as transmitted signal light; an optical receiver (20) that receives the transmitted signal light from the optical transmitter (10) via an optical fiber (40); and an optical fiber deterioration monitoring device (30) that converts backscattered light of the transmitted signal light from the optical transmitter (10) caused by the optical fiber into an electrical signal, detects the monitoring swept-frequency signal as a reference signal, and performs arithmetic processing to obtain an impulse response.

The invention relates to a digital filter circuit for calculating an exponential variance of a signal, a corresponding system on chip and a method of operation. A digital signal processing method includes: applying a first infinite impulse response filtering operation to a digital input signal to produce a first filtered signal; performing a mathematical transformation on a combination of the digital input signal and the first filtered signal to produce a transformed signal; applying a second infinite impulse response filtering operation to the transformed signal to produce a second filtered signal; performing a scaling operation on the second filtered signal to generate a scaled signal; and combining the second filtered signal and the scaled signal to produce a digital output signal indicative of a statistical characteristic of the digital input signal.

Systems, methods, and devices for combining narrowband and ultra-wideband (UWB) communication to provide enhanced ranging are disclosed. In an example aspect, a method of ranging is disclosed. In some embodiments, the method includes generating a coarse distance estimate using narrowband communications. The method may further include generating a channel impulse response (CIR) estimate using UWB communications; and estimating a distance between a first device and a second device based on the coarse distance estimate and the CIR estimate.

Various aspects of the present disclosure generally relate to wireless communication. In some aspects, a responding device may receive a signal from an initiating device. The responding device may estimate, from the signal, a channel impulse response (CIR) that represents signal reflections from one or more objects as multiple taps. The responding device may select one or more taps, from the multiple taps, that are within a first time window that starts at a first offset from a reference point and that has a first specified time duration. The responding device may transmit, to the initiating device, a CIR report that indicates the one or more taps. Numerous other aspects are described.