56 results about "Inverse filter" patented technology

The invention relates to the technical field of response time testing of electro-hydraulic proportional valves, and discloses a response time testing method and system of an electro-hydraulic proportional valve, and the method comprises the steps: collecting multi-point pressure, step response, pipeline length and medium parameters, and unifying the time reference to obtain an original pressure sequence; constructing an inverse filter spectrum to remove damping interference so as to recover a sharp peak value; lagging offset is calculated according to pipeline and medium parameters, time delay compensation is carried out, and an alignment sequence is obtained; intercepting a non-stationary segment to eliminate residual delay deviation; constructing a response dynamic data set, extracting harmonic frequency domain representation, and reconstructing a pressure sequence through inverse transformation and superposition of compensation factors; and calculating a deviation integral quantity, adjusting a dynamic compensation parameter, performing weighted correction, and outputting an optimized reconstruction response sequence. According to the method, accurate reconstruction of the real pressure response of the proportional valve can be realized, so that the accuracy and the reliability of response time testing are improved.

The invention discloses a phase recovery method based on non-interference coding aperture holography, and belongs to the field of computational optical imaging. According to the technology, phase information of an object is directly extracted from a single hologram through a physically-driven numerical optimization framework in combination with Wiener inverse filtering and an improved G-S (Gerchberg-Saxton) algorithm. A frequency domain inverse filter is constructed based on system modeling of a point diffusion hologram, noise interference is effectively suppressed, meanwhile, a historical gradient acceleration strategy is provided, a convergence path is optimized by dynamically adjusting an iteration step length, and the calculation efficiency is improved. The technology has the core advantages that the limitation of a traditional interference light path is broken through, precise optical alignment or multi-time out-of-focus surface data acquisition is not needed, the system architecture is simplified, the environmental adaptability is enhanced, and quantitative phase distribution reconstruction of transparent and weak scattering samples can be realized through single exposure.

The invention discloses a broadband noise generation method based on predistortion calibration, and relates to the technical field of communication, and the method comprises the steps: S1, setting a flatness threshold value, and screening out an original noise signal needing to be calibrated; s2, generating an inverse filter coefficient and a filter suppression coefficient for the original noise signal; s3, performing frequency response compensation by using an inverse filter coefficient, performing band-pass filtering by using a filter suppression coefficient, performing down-conversion to a baseband, and reducing a sampling rate; and S4, comparing the processed signal with a flatness threshold value, and if the processed signal does not exceed the flatness threshold value, indicating that signal calibration is completed, otherwise, entering calibration again. By introducing a pre-distortion calibration mechanism, the inverse filter coefficient is calculated in real time to compensate the system frequency response, the spectrum distortion caused by a fixed filter is effectively eliminated, and the method has the advantages of realizing dynamic and adaptive filter adjustment, effectively reducing the manual participation frequency and improving the continuous operation capability and the test efficiency of the system.

Methods and devices are provided for analyzing a tubular structure including at least two electromagnetic-acoustic transducers (EMAT) and, called sensors, attachable or attached in, on or in the vicinity of the tubular structure; and computation and / or memory resources, that are accessed locally and / or remotely and that are configured to determine, for the pair of sensors, a function representing the impulse response of the tubular structure on the basis of the diffuse acousto-elastic noise present in the structure. Developments describe the use of rings supporting the sensors; translation and / or rotation movements; permanent or temporary installations; hinged rings; various computation modes, e.g., intercorrelation, a passive inverse filter, or correlation of the coda of the correlation; the use of artificial noise sources, imaging (e.g., tomography) for determining the existence of one or more defects in the structure. Software aspects are described.

The invention discloses a parametric array loudspeaker nonlinear distortion modeling and compensation method based on deep learning. The method comprises the following specific steps: (1) recording an audio sound signal output by a parametric array loudspeaker system and an input signal of the audio sound signal at a listening position where nonlinear distortion needs to be suppressed, and making a data set for modeling; (2) performing overall modeling on the parametric array loudspeaker system by using a neural network F, and training the neural network F by using the data set; modeling a linear part of the system by using an FIR (Finite Impulse Response) filter model Flin; (3) using the trained neural network F and the model Flin to train a neural network G as an inverse filter of the system; and (4) preprocessing an audio signal expected to be replayed through the neural network G, and inputting the preprocessed audio signal into a parametric array loudspeaker system to realize nonlinear compensation of the system. According to the method, accurate modeling and effective compensation can be carried out on the nonlinear distortion of the parametric array loudspeaker, and the nonlinear distortion of the parametric array loudspeaker is remarkably inhibited.

The invention discloses a microscopic imaging device and method with a depth reconstruction effect, and the method comprises the steps: carrying out the image collection of a to-be-detected object through an image collection part, and obtaining a first image and a second image of the to-be-detected object; according to inverse filters corresponding to a plurality of preset object distances of the first shooting component and the second shooting component and a transformation matrix between the first shooting component and the second shooting component under each object distance, obtaining an offset loss amount of any pixel point under the corresponding object distance; and the depth value of the pixel point is obtained according to the offset loss amount corresponding to all the object distances of any pixel point, so that depth reconstruction is completed, and an ultra-depth-of-field image and a depth map are obtained. By adding the specially designed phase mask sheet, the optical super depth-of-field effect is realized; multiple times of shooting or use of a displacement platform is not needed in the depth reconstruction process, the ultra-depth-of-field image and the depth image can be obtained only through one-time shooting of the double cameras, and the method is simpler, more efficient and easy to implement.

The invention relates to the technical field of rotating machine fault diagnosis, and discloses a rotating machine fault diagnosis method based on adaptive blind filtering and a related device, which are used for adaptive blind filtering processing of vibration signals and enhancing the adaptive capability of fault diagnosis. The rotating machine fault diagnosis method based on adaptive blind filtering comprises the following steps: performing multi-scale band-pass filtering, and performing fault information frequency domain distribution estimation on each narrow-band signal to obtain a fault feature statistical index corresponding to each narrow-band signal; constructing a full frequency domain weighting function as a frequency domain inverse filter based on the fault feature statistical indexes corresponding to all the narrow-band signals, wherein the filter length of the frequency domain inverse filter is consistent with the Fourier transform result length of the vibration acceleration signal; performing Fourier transform on the vibration acceleration signal, performing signal processing through a frequency domain inverse filter, and performing Fourier transform to output an adaptive blind filtering signal; and performing fault diagnosis on the adaptive blind filtering signal to obtain a fault diagnosis result.

This invention discloses a machine vision-based method and system for identifying defects in wire ropes of port cranes, belonging to the field of visual defect recognition technology. The method includes: real-time acquisition of wire rope images; real-time acquisition of the wire rope's motion state data, including acceleration and angular velocity, via an inertial measurement unit; prediction of the wire rope's current motion state using a Kalman filter algorithm based on the previously acquired motion state data; construction of a fuzzy kernel function based on the predicted current motion state, and sharpening of the currently acquired wire rope image using an inverse filtering algorithm; and defect identification using a target detection model on the sharpened image. This invention enables rapid and accurate detection of surface defects in wire ropes under dynamic fuzzy and complex environments.

An electronic device according to various embodiments disclosed in the present document may comprise a communication circuit, a memory, and a processor operatively connected to the communication circuit and the memory. The memory can store instructions that, when executed by the processor, enable the electronic device to: receive, through the communication circuit, two or more pieces of audio data simultaneously obtained by one or more external electronic devices; use user information acquired from the memory so as to identify user personal HRTF information acquired on the basis of reference HRTF information; identify sound source generation location information of the two or more pieces of audio data on the basis of the two or more pieces of audio data; perform inverse filtering on the two or more pieces of audio data on the basis of the user HRFT information so as to generate an audio signal; and transmit the audio signal and the sound source generation location information to another electronic device through the communication circuit.

The invention relates to the technical field of waveform display, in particular to an eye pattern generation method and system based on a high-resolution oscilloscope, and the method comprises the following steps: analyzing a channel impact response, extracting effective delay, defining an operation kernel range, constructing sparse inverse filtering compensation distortion, judging an edge generation eye pattern track, and quantifying noise as a weight to carry out sub-pixel diffusion. According to the method, a delay set is established, a non-linear participation range is made to fit real channel behaviors, a high-order cross product is constructed in an original waveform, detail compensation of amplitude and time is enhanced in a sparse inverse filtering mode, and zero point and amplitude characteristics of a second derivative are combined, so that the error rate contour is deduced. A real edge interval is separated, continuous diffusion is formed in a histogram through a discrete probability weight constructed by quantizing noise, so that a statistical form of a low-probability region is completely reserved, and an error code related boundary presents a regular distribution structure in combination with a tail probability gradient.

This invention relates to the field of track detection technology, and discloses a method, system, device, and medium for track elevation detection. The method includes: acquiring vehicle body state data and corresponding track elevation information when a vehicle is running on different tracks; constructing an inverse filter based on a finite impulse response (FIR) compensation filter, obtaining the mapping relationship between the corresponding vehicle body state data and track elevation information through the inverse filter, and optimizing the delay of the inverse filter using a recursive least squares method based on the mapping relationship; optimizing the order of the inverse filter using power spectral density analysis; reconstructing the acquired vehicle body response data based on the optimized delay and order of the inverse filter; training a convolutional neural network using all reconstructed vehicle body response data and corresponding track elevation information to obtain a track detection model, which is then used to detect the elevation of a target track, thereby improving the accuracy of track elevation detection.

The invention discloses a method for enhancing DTS spatial resolution based on inverse filtering. The method comprises the following steps that firstly, observation signals are collected and preprocessed; establishing an observation model, representing the preprocessed observation signal as convolution of real temperature distribution and system detection pulse in a spatial domain, and superposing noise; then discrete sequence representation of the detection pulse is obtained or constructed, and normalization processing is carried out; constructing a regularization inverse filter in a frequency domain based on the detection pulse and the preprocessed observation signal; performing frequency domain transformation on the preprocessed observation signal, and multiplying the frequency spectrum of the observation signal by a regularization inverse filter in a frequency domain to obtain a recovered frequency domain signal; and carrying out inverse frequency domain transformation on the recovered frequency domain signal, and obtaining a time domain recovered signal after taking a real part. The method can effectively recover the fuzzy temperature signal characteristics of the detector, improves the reduction precision of the peak value and the edge, introduces the regularization term to suppress the high-frequency noise amplification, and gives consideration to the recovery performance and the stability.

The embodiment of the invention provides a method for processing an audio signal, electronic equipment and a vehicle. The method comprises the following steps: determining an impact response function of a plurality of audio input modules in a vehicle, wherein the impact response function is used for representing an acoustic transmission path from a sound source to the audio input modules; determining an inverse filter function of a corresponding audio input module in the plurality of audio input modules according to the impact response function; determining a human voice enhancement signal and a noise reference signal according to the inverse filter function; and performing adaptive filtering on an audio signal input to the audio input module according to the human voice enhancement signal and the noise reference signal. According to the method disclosed by the embodiment of the invention, the voice enhancement performance can be remarkably improved by using the acoustic environment of the vehicle with a relatively low calculation amount, so that the user experience is improved.

The application discloses a kind of parametric array loudspeaker nonlinear distortion modeling and compensation method based on deep learning.The specific steps are as follows:(1) record the audio sound signal and its input signal of parametric array loudspeaker system output at the listening position where nonlinear distortion needs to be suppressed, and prepare the data set for modeling;(2) use neural network F to model the whole parametric array loudspeaker system, train neural network F using the data set;And using FIR filter model F lin Model the linear part of the system;(3) use trained neural network F and model F lin Train neural network G as the inverse filter of the system;(4) input the audio signal expected to be played into parametric array loudspeaker system after pre-processing by neural network G, to realize compensation for the nonlinear of the system.The method of the application can accurately model and effectively compensate the nonlinear distortion of parametric array loudspeaker, and significantly suppress the nonlinear distortion of parametric array loudspeaker.

The present invention relates to the technical field of wind power generation equipment monitoring, and specifically relates to a method and device for health monitoring of an offshore wind power tower barrel and a storage medium. The method includes: collecting environmental noise and recording it as a time-domain signal, segmenting the time-domain signal to form multiple segments of time-domain signals; preprocessing all the time-domain signals; performing passive inverse filtering on the preprocessed time-domain signals to obtain a time-domain impulse response matrix; analyzing the modal coupling characteristics of the tower barrel, drawing a three-dimensional wave velocity distribution map, and performing health monitoring on the tower barrel. The present invention uses environmental noise caused by wave impact as an excitation source, without manual intervention. For the collected environmental noise, by combining time-domain, frequency-domain, and spatial-domain features, a time-domain impulse response matrix is finally obtained, which is beneficial to analyzing the overall modal coupling characteristics of the tower barrel and drawing a high-precision three-dimensional wave velocity distribution map, thereby achieving sub-millimeter detection accuracy and reducing the false alarm rate.

The invention discloses a chaotic image encryption method based on an attention mechanism and a filter, and the method comprises the steps: inputting and preprocessing, self-attention feature extraction and key generation, chaotic system initialization and sequence generation, DNA coding encryption, and dynamic three-dimensional filtering diffusion. All parameters and sequences need to be reproduced by using the same key vector, inverse permutation, inverse filtering diffusion and DNA decoding are carried out in sequence, and finally an original image is subjected to lossless recovery. The invention relates to the technical field of information security, and in the method, a secret key is dynamically generated from image content through a self-attention mechanism, and a dynamic three-dimensional filtering diffusion mechanism is adopted to realize collaborative encryption and dynamic randomization processing of a color image three-dimensional structure, so that the security, diffusivity and anti-attack capability of an encryption system are remarkably improved.

A sound propagation characteristics correction apparatus includes: a pseudo inverse filter generation unit that generates a pseudo inverse filter by a function that smooths sound propagation characteristics from a speaker driver of a wearable open-ear speaker to an ear canal entrance; and an acoustic signal correction unit that corrects an acoustic signal on the basis of the pseudo inverse filter and transmits the corrected acoustic signal to the wearable open-ear speaker.

The present invention relates to an ultra-directional speaker system comprising: a first envelope calculation unit for calculating the envelope of the currently input audio input signal; a square root calculation unit for generating a square root signal of a first envelope signal by calculating a square root of the first envelope signal calculated by the first envelope calculation unit; a pre-distortion adaptive filter unit for generating a compensation signal by performing distortion compensation in which an adaptive filter coefficient update term according to an adaptive filter coefficient determined in a previous stage is applied to the currently input audio input signal; a second envelope calculation unit for generating a second envelope signal by calculating the envelope of the compensation signal; an error calculation unit for generating an error signal by comparing and calculating the second envelope signal with the square root signal of the first envelope signal; an adaptive filter coefficient update unit for calculating an adaptive filter coefficient update term and an adaptive filter coefficient from the error signal; a dynamic signal modulation unit for generating a modulation signal by dynamically modulating the compensation signal into an ultrasonic band; an ultrasonic transducer model for modeling an inverse filter corresponding to frequency characteristics of an ultrasonic transducer, and generating a filtering signal by applying the inverse filter to the modulation signal; an ultrasonic amplifier for amplifying the filtering signal; and the ultrasonic transducer for converting the amplified signal into an ultrasonic signal, wherein the dynamic signal modulation unit includes: an AM signal modulation unit for generating an AM modulation signal of transmission data; a first sideband signal modulation unit, which receives the AM modulation signal and performs a Hilbert operation thereon so as to obtain a first sideband modulation signal from which a carrier wave has been removed; and a second sideband signal unit, which receives the first sideband modulation signal and adds the carrier wave thereto so as to obtain a second sideband modulation signal.

The application relates to a microcomputer video information protection method and system based on variable sequence related interference, which specifically comprises the following steps: collecting an original video signal and calculating a video timing parameter; configuring a multiple clock source frequency according to a line frequency, generating and dynamically rearranging an initial pseudo-random sequence to obtain a variable sequence related interference signal; identifying a scanning interval, generating a digital scrambling pulse during a forward stroke and superimposing the variable sequence related interference signal to generate a time domain scrambling composite signal, and attenuating the variable sequence related interference signal to generate the time domain scrambling composite signal during a reverse stroke; generating a wideband white noise signal, mixing the wideband white noise signal with the time domain scrambling composite signal to obtain a full-band mixed interference signal; generating an envelope modulation waveform, and amplitude modulating the full-band mixed interference signal to generate a simulation video radiation interference signal; collecting a real-time radiation intensity adjustment gain and outputting a target interference emission signal; collecting a power line conduction leakage signal, and generating an inverse filtering signal to offset the power line conduction leakage signal. The application improves the security of microcomputer video information.

The invention discloses a medium-voltage distribution network fault distance measurement method and system based on traveling wave waveform reduction, and relates to the technical field of distribution network distance measurement, and the method comprises the steps: building a total transmission function which reflects a traveling wave signal multistage distribution transformer according to the collected entrance capacitance, line impedance and traveling wave transmission time delay of a to-be-detected power distribution network line; introducing a set frequency domain compensation mechanism into the total transmission function to obtain a target inverse filtering compensation function; the target inverse filtering compensation function is converted into a discrete problem, and regularization correction and parameter optimization are carried out in sequence; solving a regularization equation obtained after optimization, and restoring an original traveling wave signal waveform according to a solving result; and realizing high-precision fault distance measurement of the to-be-detected power distribution network line according to the original traveling wave signal waveform.

This invention relates to the field of response time testing technology for electro-hydraulic proportional valves, and discloses a method and system for testing the response time of electro-hydraulic proportional valves. The method includes: acquiring multi-point pressure, step response, pipeline length, and medium parameters, and unifying the time base to obtain an original pressure sequence; constructing an inverse filter spectrum to remove damping interference and recover sharp peaks; calculating hysteresis offset based on pipeline and medium parameters to perform time delay compensation and obtain an aligned sequence; truncating non-stationary segments to eliminate residual delay deviations; constructing a dynamic response dataset and extracting harmonic frequency domain representations, and reconstructing the pressure sequence by inverse transformation and superimposing compensation factors; calculating the deviation integral, adjusting dynamic compensation parameters and weighting corrections, and outputting an optimized reconstructed response sequence. This method can achieve accurate reconstruction of the true pressure response of the proportional valve, thereby improving the accuracy and reliability of response time testing.

Provided is a playback sound correction apparatus that corrects a playback sound of an open-ear earphone including a speaker and a microphone, the apparatus including: a characteristics measuring sound playback unit configured to play back a characteristics measuring sound from the speaker; a transfer function calculation unit configured to calculate a transfer function from the speaker to the microphone based on a recorded sound of the microphone and the characteristics measuring sound; a correction filter calculation unit configured to calculate an inverse filter of the transfer function as a correction filter; and a playback sound correction unit configured to correct the playback sound with the correction filter.

The invention provides a positioning method, device and equipment based on a square root inverse filter, and a storage medium, and the method comprises the steps: obtaining first measurement data of a GNSS receiver, second measurement data of an IMU sensor, and third measurement data of a camera through constructing a basic structure of the square root inverse filter, carrying out the preprocessing of the second measurement data, and updating the IMU state. According to the updated IMU state and the first measurement data, reference frame transformation initialization is carried out to determine a transformation frame, marginalization processing is carried out on the third measurement data to obtain a marginalization processing result, and through a square root inverse filter, the marginalization processing result, the first measurement data and the transformation frame, the first measurement data is obtained. And performing joint updating and iteration to determine a positioning result, realizing tight coupling of the first measurement data, the second measurement data and the third measurement data based on the square root inverse filter, and improving the accuracy of the positioning result.

The application relates to the technical field of speech recognition, in particular to a speech-based emotion recognition method, which comprises the following steps: according to an input speech signal, frame division and windowing are carried out. The application can obtain more profound insights into emotional expression by decomposing the input speech signal into two physical sources of glottal excitation and vocal tract response for independent modeling and analysis, obtaining a vocal tract transfer function set via linear predictive coding operation, and applying inverse filtering to the original speech signal to reconstruct an approximate glottal pulse sequence, effectively stripping the influence of vocal tract resonance on the signal, so that perturbation parameters, open quotient and closed quotient and the like representing vocal cord vibration patterns can be directly calculated, at the same time, the vocal tract transfer function set is used to identify and track the dynamic trajectory of the formant, and the phase difference cosine mean between adjacent frames is combined to quantify the sound production stability, and subtle dynamic adjustment and control stability of sound production organs such as the oral cavity and tongue position caused by emotional changes are captured.

The invention relates to a bandwidth limitation and electric dispersion cooperative pre-compensation method based on an improved GS algorithm, and belongs to the technical field of optical fiber communication and compensation. According to the method, a filter for balancing bandwidth limitation is designed and integrated into an improved GS pre-EDC method, and the method specifically comprises the steps that 1, the improved GS pre-EDC is adopted at a transmitting end, balancing is conducted after FFE is used at a receiving end, and the tap coefficient of the FFE is stored as hBL; 2, initializing a signal; step 3, executing receiving end constraint on the amplitude of the odd number samples of the signal; step 4, converting the signal to a transmitting end through an inverse dispersion frequency domain transmission function and an inverse filter; step 5, executing sending end constraint on the phase of the signal; step 6, converting the signal back to a receiving end by using a dispersion frequency domain transmission function and a filter; and step 7, repeating the steps 3-6 to complete N iterations. The pre-compensation performance of the method is superior to that of an improved GS pre-EDC method, the bandwidth requirement for a photoelectric device in a high-speed system is reduced, and post-equalization is not needed.

The present invention discloses a method for obtaining an ideal delta pulse excitation using a high-frequency pulse sound source, comprising the steps of: recording a high-frequency electric spark pulse sound source in a fully anechoic chamber and recording the signal as a WAV file; performing inverse filtering on the WAV file and saving the obtained inverse filter as a WAV file; performing a binaural impulse response test using the high-frequency electric spark pulse sound source and a scaled artificial head in a scaled model, and saving the scaled binaural impulse response file as a WAV file; generating n binaural impulse responses that are approximate to those obtained when the ideal delta pulse signal is used as the sound source; performing scale ratio conversion to obtain the binaural impulse response corresponding to the full-scale hall; and performing phase-locked averaging to obtain a binaural impulse response that can be used in subjective listening experiments. The present invention can obtain a binaural impulse response that is approximate to that obtained when the ideal delta pulse signal is used as the sound source, further improve the bandwidth and accuracy, and simultaneously improve the signal-to-noise ratio, making the obtained signal more suitable for auralization experiments.

A controller calculates individual parameters, including nonlinear parameters of a speaker model, based on a measured displacement of a vibration system of a speaker. A sound generator outputs an audio signal, and an output flattening filter adjusts gains of individual bands of an audio signal so that a frequency characteristic of a volume of sound output by the speaker is flattened in accordance with a speaker model obtained by linearizing calculated parameters and outputs a resultant audio signal. A nonlinear inverse filter corrects the resultant audio signal by signal processing adapted to the gains of the output flattening filter so that the nonlinear distortion of the speaker is suppressed in accordance with the speaker model including the nonlinear parameters and outputs the corrected audio signal as an output audio signal to the speaker via an amplifier.

Implementations of the disclosure provide a video processing method for invertible filtering in video coding. The video processing method may include receiving, by a processor, an image derived from encoded video information stored in a bitstream. The video processing method may also include applying, by the processor, a forward filter to the image to generate an intermediate image and flipping the intermediate image. The video processing method may further include applying, by the processor, an inverse filter to the flipped intermediate image to generate a reconstructed image, and inversely flipping the reconstructed image to generate an invertible reconstructed image.

An ear-wearable device includes a receiver that outputs an acoustic signal in response to a drive signal, a source device that provides a reference audio signal, and an inward facing microphone that converts in-ear acoustic sound to an error signal. The device includes an active noise controller that provides a noise control signal based on the error signal and a processing block that provides an amplified hearing signal based on the reference audio signal. The device includes an instability detector configured to: determine a transform function that represents an approximation of an inverse of the active noise controller; apply the transform function to the drive signal to determine an inverse filter signal; apply a filter to the error signal. The filter compensates for deficiencies in the transform function. The active noise controller is disabled when an anomalous similarity between filtered error signal and the inverse filter signal is detected.

A speaker has a magnetic angle sensor that detects the displacement of the vibration system of the speaker. A non-linearity compensation filter compensates for output distortion due to a non-linear parameter of the speaker. A transfer function of a linear inverse filter is applied so that output distortion is eliminated from the displacement, detected by the magnetic angle sensor, of the vibration system. A controller drives the speaker by using a test signal, and measures a response from the detected displacement of the vibration system. If error between the measured response and the theoretical value, theoretically determined from the design specifications, of the response is small, the controller uses the theoretical value, theoretically determined from the design specifications, of the non-linear parameter as an active non-linear parameter and sets, in the non-linearity compensation filter, a transfer function that compensates for distortion in an output due to the active non-linear parameter.