61 results about "Amplitude response" patented technology

The invention provides a TMD lockout control method for motion suppression of a semi-submersible platform, and the method comprises the steps: carrying out the frequency domain decomposition of a random wave predicted by a target sea condition, obtaining the energy distribution of each frequency component, obtaining the equivalent amplitude of each frequency component according to the energy distribution, considering the amplitude response of the platform under the action of waves of different frequencies, and achieving the motion suppression of the semi-submersible platform. The method comprises the following steps of: calculating a weight ratio of platform motion response caused by each frequency component under the action of random waves by integrating three types of information, namely amplitude, amplitude response and motion suppression effect, and finally carrying out weighted combination on the locking time corresponding to each frequency component according to the weight ratio, so as to obtain the motion suppression effect of the platform. And the optimal locking control time suitable for the current random wave is obtained. According to the method, a high-complexity non-causal control command does not need to be deduced, control parameters can be updated in real time according to short-term sea condition changes, pitching response of a platform is effectively reduced, and comprehensive optimization control over multi-frequency waves is achieved.

Methods and systems for generating a sound. The method includes obtaining meshes representing at least one object in a frame of an environment, the environment including a source and a receiver; determining spatial continuity information and reflection normal information of the meshes; determining, based on the spatial continuity information and the reflection normal information, reflection paths between the source and the receiver involving the at least one object, each of the reflection paths having at least one reflection point associated with the at least one object; obtaining spatially sampled results by spatially sampling a space around the at least one reflection point of reflection paths using multiple distributions of rays, the spatially sampled results correlating with geometric information of the meshes; generating reflection amplitude responses for each of multiple audible frequencies in the environment based on the spatially sampled results; and producing a sound based on the reflection amplitude responses.

The invention discloses an unmanned aerial vehicle noise elimination method and device for cross-medium direct acoustic communication, and the method comprises the steps: carrying out the sampling of a known detection signal and a receiving signal, obtaining a frequency domain expression, calculating a cross-medium sound pressure transmission function, and extracting amplitude response, phase response and group delay; based on a normalization ratio and local maximum detection method, identifying a center frequency set of unmanned aerial vehicle propeller noise; the notch depth, bandwidth, frequency compensation and phase compensation are determined by combining channel characteristic parameters, and a cascade notch filter is constructed; and inputting the channel characteristic parameters into the deep learning model, outputting a filter parameter correction amount, realizing dynamic updating of the filter, obtaining the filter capable of suppressing noise, and further obtaining a denoised communication signal. According to the invention, precise detection of propeller noise frequency and multi-dimensional adaptive adjustment of filter parameters can be realized.

The invention is suitable for the technical field of geophysical exploration, and provides a rock physics-driven time-frequency domain longitudinal wave velocity dispersion attribute prediction method, which comprises the following steps of: establishing a rock physics model; under the driving of the rock physical model, inverting the porosity of the microfractures in the well, calculating the skeleton elastic modulus of the rock containing the fractures, and predicting a frequency-dependent longitudinal wave velocity logging curve; constructing a time-frequency domain longitudinal wave velocity seismic inversion initial model; constructing a TF-AVO theoretical formula, developing a time-frequency domain longitudinal wave velocity seismic inversion method based on an L-BFGS algorithm, and performing longitudinal wave velocity inversion by taking the initial model as prior information; analyzing the relative change of the frequency-varying longitudinal wave speed under different frequencies, and quantitatively predicting the frequency dispersion attribute of the longitudinal wave speed; and on the basis of logging data, calibrating a longitudinal wave velocity dispersion attribute prediction result, and carrying out reservoir oil and gas detection. The method has higher precision and resolution, the stability and reliability of longitudinal wave velocity dispersion attribute prediction are remarkably improved, and the dependence on strong amplitude response is overcome.

The invention discloses a millimeter wave dual-polarization phased-array antenna beam forming method and device, and relates to the technical field of high-frequency communication, and the method comprises the following steps: obtaining a complex weight signal of each array element in an antenna array under a horizontal polarization channel and a vertical polarization channel, respectively carrying out amplitude response and phase response acquisition and calibration processing on the complex weight signal; performing dynamic extraction and modeling on amplitude errors and phase errors of each array element in different polarization channels; on the basis of error estimation, constructing a machine learning prediction module comprising a time sequence prediction model; performing joint estimation on a real-time error estimation result and a prediction error result; constructing a hardware-software collaborative calibration structure comprising a dynamic gain control module on the basis of the compensation weight; beam focusing of the target space area is realized; according to the invention, the hidden problems of beam offset, sidelobe lifting, gain loss and the like caused by inconsistency of polarization channels are solved.

The invention discloses a direct-current system ground fault real-time monitoring method and system, and relates to the field of intelligent power distribution, and the method comprises the steps: collecting the bus voltage, branch current and disturbance response signals of each branch, constructing a fusion abnormal factor in a fixed monitoring window, comprehensively reflecting the voltage change rate, disturbance amplitude response and current vector offset characteristics, and carrying out the real-time monitoring of the ground fault of a direct-current system. And inputting the factor and a disturbance parameter into a nonlinear scoring function together, and outputting a grounding probability confidence score. And for the branch with the scoring result exceeding a set threshold value, the system executes bypass closing and main branch opening operation, voltage and current derivative data are collected in a recovery window, the voltage drop recovery rate is calculated, and whether an actual grounding point is formed or not is judged. And if the fault is confirmed to be true, the isolation state is maintained, and branch reset operation is completed through a recovery criterion and a manual permission mechanism in a subsequent period. The method has better effects in real-time performance, accuracy and safety.

A method and system for bandwidth extension of a critical sampling circuit, belonging to the field of signal sampling technology. The implementation method is as follows: S1, estimating the model coefficients of the squared amplitude response of the simulated channel of the critical sampling circuit through ultra-relaxed primordial duality, constructing the simulated channel response matrix, and obtaining the system response matrix by row decimation; S2, establishing the high-dimensional response matrix of the critical sampling circuit and completing parameter initialization; S3, performing iterative calculation of the extension features through elastic network regression, checking the convergence of the extension residual or the number of iterations, if satisfied, proceeding to S4, otherwise continuing iteration; S4, reconstructing the high-dimensional signal using the extension features calculated in S3, and obtaining the reconstructed signal after bandwidth extension of the critical sampling circuit through sampling rate conversion. This invention, by iteratively calculating the spectral characteristics within the target extended bandwidth of the critical sampling circuit, can recover out-of-band lost spectral information without increasing the sampling clock frequency and analog input bandwidth, achieving low-cost high-frequency signal sampling.

The invention discloses a pedestal pan mute control method and device and a pedestal pan, and relates to the technical field of pedestal pan control, and the method comprises the following steps: in response to a mute signal, calculating a mute amplitude coefficient according to the current state data of the pedestal pan; according to the mute amplitude coefficient and the initial action amplitude of the pedestal pan, the mute action amplitude of the pedestal pan is obtained; and in response to the control signal, executing a toilet action corresponding to the control signal according to the toilet mute action range. The method has the beneficial effects that the mute amplitude coefficient is calculated according to the current state data of the pedestal pan, so that the mute amplitude is changed along with the change of the using state, the cover opening state and other states of the pedestal pan, and thus actions of different amplitudes are executed in different states of the pedestal pan; and therefore, a higher cleaning effect can be achieved under the condition of being used by people for a long time, and both the cleaning effect and the mute effect are achieved.

Various current measurement systems and methods of signal processing are disclosed. In one example, there is a current measurement system that includes a filter with a corner frequency within the operating frequency range of the current measurement system. This provides a system with good SNR, whilst also preventing or reducing the likelihood of output saturation during large di / dt spikes. The system further includes an equaliser arranged to compensate for the phase and / or magnitude response of the filter within the operating frequency range.

The embodiment of the invention provides a parameter determination method and device, electronic equipment, a chip and a medium, and relates to the technical field of audio signal processing, and the method comprises the steps: obtaining first amplitude response data corresponding to a first processing unit in a plurality of processing units of a signal processing device; based on the first amplitude response data corresponding to the first processing unit, respectively determining weights of a plurality of processing units of the signal processing device; and determining a first target gain of the first processing unit based on the weight and an initial gain of the signal processing device, the first target gain being used for determining a first frequency response control parameter corresponding to the first processing unit. According to the method provided by the invention, the gain of each processing unit is automatically calibrated by using iterative computation, so that the accurate control of interaction between the processing units in amplitude-frequency response is realized, the overall precision of the signal processing device is improved, the gain adjustment and the improvement of the signal processing efficiency are further realized, and the accuracy of the signal processing device is improved. And the manual debugging cost is reduced.

The invention discloses a shallow lake turbidity remote sensing inversion method and system based on wind drive physical constraint, and the method comprises the steps: constructing a time sequence pairing sample through combining a remote sensing image and actually measured data, and extracting wind drive characteristics such as wind speed, wind stress and strong wind duration; establishing direction consistency priori by using monotone calibration to generate an expected direction signal, and constructing a wind-driven amplitude response surface interpolation to obtain an expected change amplitude; in model training, a double-moment regression network sharing weights is adopted, and an actual measurement supervision error and wind-driven physical constraints are jointly optimized; and finally, carrying out pixel-level reasoning and verification on the multi-temporal remote sensing image to realize continuous estimation of spatial distribution and dynamic change of turbidity. According to the method, a double-moment input and shared weight network structure is introduced, the physical mechanism and deep learning advantages are fused, the time sequence continuity, the physical interpretability and the cross-space-time generalization ability are remarkably improved while high inversion precision is kept, and an efficient and reliable technical approach is provided for water environment remote sensing monitoring.

An antenna tuner receives outputs of detectors coupled to an input and / or an output of each power amplifier (PA) of a plurality of PAs of a multiple antenna system. A processor of the antenna tuner is configured to process the detector outputs and to generate at least one of a gm gate bias value and a cascode gate bias value to be applied to gm and cascode gates, respectively, of a last stage of each of the PAs to change the shape of an input amplitude-to-output amplitude (AM-AM) response of each of the PAs independently of one another. The processor can also be configured to process the detector outputs to generate a varactor bias value to be applied to a terminal of a varactor of each of the PAs to change the shape of an input amplitude-to-output phase shift (AM-PM) response of each of the PAs independently of one another.

The invention relates to an ultrahigh frequency sensor sensitivity checking system and method based on program control analog pulses, and belongs to the technical field of power equipment detection. The system comprises an analog pulse generation device which outputs a program-controlled pulse sequence with controlled parameters including pulse amplitude, width, rise time and oscillation frequency to a transformer; the ultrahigh-frequency sensor to be detected detects a program-controlled pulse sequence induced by the transformer and outputs a detection pulse sequence corresponding to the program-controlled pulse sequence; the sensitivity checking unit is respectively connected with the analog pulse generation device and the ultrahigh frequency sensor to be detected, and is used for receiving and analyzing the amplitude-frequency characteristics of the program-controlled pulse sequence and the detection pulse sequence; and determining the minimum detection sensitivity of the to-be-checked ultrahigh frequency sensor through amplitude response characteristic analysis, and determining the detection frequency band range of the to-be-checked ultrahigh frequency sensor through frequency response characteristic analysis. According to the invention, flexible and accurate verification of the sensitivity of the ultrahigh frequency sensor is realized.

The invention relates to the technical field of power equipment state evaluation, and discloses a three-phase transformer state evaluation method, system, equipment and medium, and the method comprises the steps: collecting the frequency-amplitude response data of each phase winding of a three-phase transformer; dividing at least two characteristic sensitive frequency bands, wherein the frequency bands respectively correspond to core components of the transformer; extracting difference characterization characteristic parameters of each phase and each frequency band, and calculating a three-phase imbalance coefficient; and generating a dynamic threshold value based on the operation condition of the transformer and historical data, performing multi-level state evaluation in combination with the deviation degree of the unbalance coefficient and the threshold value, and positioning an abnormal phase. The method breaks through the limitation of a traditional fixed threshold value and a single feature, adapts to different operation scenes through the dynamic threshold value, expands feature parameter types and frequency band configuration flexibility, achieves state accurate grading and anomaly tracing, solves the problems that a traditional method is poor in adaptability and narrow in protection range, and remarkably improves the evaluation universality and the intelligent level.

A dynamically controllable metasurface antenna system is disclosed. The system comprises: a metasurface comprising a plurality of electronically addressable meta-atoms, each meta-atom comprising an integrated electronic component configured to control a local phase and / or amplitude response in response to an applied bias; a feed configured to launch surface waves onto the metasurface; and a control circuit operatively coupled to the plurality of meta-atoms and configured to selectively bias individual meta-atoms to dynamically control at least one of frequency response, beam direction, or polarization state of the antenna system in real time.

The invention relates to the field of radar array signal processing, in particular to a space-time joint domain beam forming method in a fast moving target scene. Firstly, a sensor array is constructed, initialization parameters including array configuration parameters, incident angles of target echo signals and interference signals, Doppler frequency shifts and the like are set, and a space-time two-dimensional steering vector model is established; combining the model, determining a beam amplitude response constraint condition according to an incident angle and a Doppler frequency shift range of a target echo signal, and constructing a space-time two-dimensional robust beam forming model based on a minimum output power criterion; obtaining a covariance matrix of the low-sample echo data, and reconstructing the covariance matrix after eigenvalue decomposition and eigenvalue compensation; and finally, a space-time adaptive processing model is constructed based on the reconstruction matrix, an initial weight vector is obtained through relaxation non-convex constraint, and a final weight vector is obtained through phase compensation iterative optimization until conditions are met, so that the robustness and accuracy of beam forming in the case of low sample data in a fast moving target scene can be improved.

The application discloses a scalable multi-array towed electromagnetic coil structure and a working method, and belongs to the field of towed transient electromagnetic detection equipment. The structure adopts a multi-array modular array design, one transmitting array element and two receiving array elements on the two sides of the transmitting array element form a modular working unit, and the compact connection of the array is realized by the common use of the receiving array elements of adjacent working units. The horizontal offset distance and the vertical distance between the transmitting array element and the receiving array element are constrained, the mutual inductance coefficient in the working unit approaches to zero, the primary field residual interference is suppressed at the physical level, and the shallow detection blind area is eliminated. Time division multiplexing technology is adopted to realize multi-unit round transmission and multi-channel signal synchronous acquisition, a spatial amplitude response matrix is constructed, and the accurate positioning of the center position and the geometric boundary of an underground abnormal body is realized. The application supports horizontal modular seamless expansion, significantly improves the scanning width and the detection efficiency, and can provide rich data basis for high-quality inversion of the underground abnormal body.

The application discloses a channel response related noise shaping method based on a pre-equalization DMT transmission system, comprising the following steps: S1: considering the influence of a channel, minimizing the difference between a sending signal and a receiving signal, and obtaining optimal feedback filter coefficients; and S2: using the optimal feedback filter coefficients to generate a quantized output signal, wherein the quantized output signal is used to be loaded into a low-quantization digital-to-analog converter to generate an analog signal. ‑1 The application considers the influence of a channel response in the process of obtaining tap coefficients of a feedback filter, and the weighting coefficient changes with the change of the channel H, so that the amplitude response abs|1+G| of the feedback filter changes with the inverse channel response H Changes are made, and unbalanced damage caused by residual quantization noise is effectively avoided. In addition, most of the residual quantization noise is concentrated in the high-frequency part of the signal, so that the residual quantization noise is further attenuated after transmission through the channel, and the SQNR is improved.

The present disclosure relates to a 5G or 6G communication system for supporting a higher data transmission rate. A hybrid MIMO system includes N digital transceiver chains and N analog beamformers, each analog beamformer coupled to one of the digital transceiver chains and including M analog TX chains and M analog RX chains. One of the simulated RX chains is an RX reference node, and one of the simulated TX chains is a TX reference node. For each beamformer, the processor measures TX and RX phase responses and TX and RX amplitude responses of TX and RX reference nodes, respectively, using a transceiver chain and a calibration transceiver; determining a reciprocity calibration based on a phase difference between the TX and RX phase responses and an amplitude difference between the TX and RX amplitude responses; and applying a reciprocity calibration to the stored beam definition to correct the phase and amplitude difference.

The embodiment of the application provides a kind of automation test control method and system of multi-channel amplitude-phase amplification module, belong to electronic test technical field.The method comprises: obtaining the amplitude response data and the phase response data of each channel in multi-channel amplitude-phase amplification module under preset test excitation condition, and constructs channel reference parameter set;Sequence influence model is constructed, and the test execution order of each channel is optimized;The amplitude-phase test is executed to each channel, the amplitude response data and the phase response data of each channel at current time are obtained, and virtual reference channel parameter is constructed;The amplitude response data and the phase response data of each channel are corrected in consistency, and the corrected amplitude-phase parameter of each channel is obtained, and the amplitude-phase test result corresponding to each channel is generated based on corrected amplitude-phase parameter.The application scheme realizes the unified alignment and error suppression of multi-channel amplitude-phase test result, improves test consistency and stability.

A correction device (100) for a multi-channel time-interleaved ADC system (300) is disclosed. The correction device (100) is configured to correct amplitude response mismatch errors of an ADC output sequence v using a first time-varying function B and further to correct phase response mismatch errors of the ADC output sequence using a second time-varying function C. Thus, due to the use of the first time-varying function B and the second time-varying function C, low complexity converter performance is improved.

The application relates to the technical field of short-wave communication and provides a 3G-ALE burst waveform identification method, device and electronic equipment based on a correlation peak fusion graph. The method comprises the following steps: S1, according to the protocol frame structure of various burst waveforms in a 3G-ALE physical layer, extracting preset fields irrelevant to service payloads in the various burst waveforms, and constructing complex baseband template sequences corresponding to the various burst waveforms; S2, obtaining a complex baseband observation sequence corresponding to a 3G-ALE burst signal to be identified; S3, performing cross-correlation calculation on the complex baseband observation sequence and the complex baseband template sequences corresponding to the various burst waveforms respectively, and obtaining correlation amplitude responses corresponding to the various burst waveforms; S4, mapping all the correlation amplitude responses to corresponding predetermined regions in a graph according to a preset spatial layout, and obtaining a correlation peak fusion graph; and S5, inputting the correlation peak fusion graph into a trained 3G-ALE burst waveform identification model, and obtaining the waveform category of the 3G-ALE burst signal.

The invention discloses a sound compensation method and device, an earphone and a storage medium, and the method comprises the steps: responding to the condition that the earphone is worn on a human ear, playing a test audio signal through a moving-iron loudspeaker and a moving-coil loudspeaker, and collecting an ear canal signal of the test audio signal through a feedback microphone; determining an amplitude response curve of a first frequency band based on the test audio signal and the ear canal signal, and determining a response error curve based on the amplitude response curve; determining time delay information of the ear canal signal based on a plurality of candidate crossover frequencies; and determining a plurality of loss values based on the plurality of candidate crossover frequencies, the response error curve and the time delay information, determining a digital filtering coefficient based on the plurality of loss values, and performing sound compensation based on the digital filtering coefficient. And the accuracy of sound restoration and the consistency of hearing feeling are obviously improved.

The application discloses a signal receiving method for spread spectrum underwater acoustic communication of long-distance time-varying channel, which comprises the following steps: obtaining the received communication signal, carrying out down-conversion and low-pass filtering to obtain a baseband signal, and carrying out synchronization to divide the baseband signal into pilot signals and spread spectrum sequence signals; using the pilot signals to complete initial channel estimation to obtain initial Doppler estimation value, time delay and amplitude response estimation value of each path; using the estimated Doppler to carry out frequency offset compensation and resampling on the first spread spectrum sequence receiving signal; using the estimated time delay and amplitude response to construct a first to-be-estimated spread spectrum sequence signal Rake receiver; combining the Rake receiver and using maximum likelihood estimation to realize the decision of the current spread spectrum sequence and the log-likelihood ratio of each bit; using the decided spread spectrum sequence to estimate the Doppler, time delay and amplitude response of the current spread spectrum sequence signal; and in this way, the decision of all spread spectrum sequences is completed. The application can effectively reduce the demodulation error rate.

The invention provides a laser light source line width stability multi-parameter joint evaluation method and system. The method comprises the following steps: acquiring junction temperature parameters of a laser, environmental sound spectrum data and optical interference original signals in a power abrupt change period; performing energy accumulation operation on the environmental sound spectrum data to generate an acoustic characteristic energy value sequence; performing phase demodulation processing on the optical interference original signal to generate a line width drift value sequence; performing frequency domain correlation operation on the junction temperature parameter, the acoustic characteristic energy value sequence and the line width drift value sequence to generate a joint transfer function matrix; calculating the amplitude response modulus length of the matrix to obtain a dynamic weight vector; performing Fourier transform on the line width drift value sequence to obtain an initial line width frequency domain distribution vector, and calculating a multi-parameter joint evaluation value in combination with the dynamic weight vector; according to the invention, dynamic weighted comprehensive evaluation of the line width stability of the laser light source is realized, and the accuracy and robustness of evaluation are improved.

The present application relates to a kind of LFMs based on amplitude-phase response measurement timing method, belong to telemetry, radar and test measurement technical field, in the measurement method of amplitude-phase response, it is usually based on the way of single tone or double tone scanning and time-consuming is longer, and single tone cannot measure system phase response;And based on the way of multi-carrier, due to power limit, leading to the signal-to-noise ratio of each sub-carrier reduces, measurement result is poor. Again due to the related advantages of LFM signal, such as bandwidth, energy concentration and pulse compression etc., it gradually obtains application in this kind of measurement calibration system. After using the timing estimation of the present application, not only the amplitude response can be accurately measured, but also the phase response can be accurately measured. The actual measurement results show that the phase measurement curve of each time is very close to even coincide, so the idea of average noise reduction can be further used to improve the measurement accuracy of phase response.

The present invention concerns a computer implemented method for processing an audio signal for which an audio signal containing speech is obtained. Then a smoothed difference signal is derived from the obtained audio signal and smoothed spectrum thereof. At least one peak is detected in the obtained difference signal and a peaking filter generated based on the at least one detected peak. The peaking filter comprises at least a Q factor and a gain parameter, whereas the Q factor is based on the geometrically obtained cut-off frequencies around the at least one detected peak; and the gain parameter is derived from a gain interaction matrix with elements based on magnitude responses at cut-off frequencies around the at least one detected peak.

The application provides a decoder LLR weight acquisition method, comprising the following steps: acquiring a noise variance estimation value of a received signal by using a difference value of channel estimation values of two adjacent symbols of the received signal; acquiring an equivalent power estimation value of the received signal by using an amplitude value of the channel estimation value; and obtaining an LLR weight based on a ratio of the equivalent power estimation value of the received signal to the noise variance estimation value. The decoder LLR weight acquisition method utilizes the similarity of channels between adjacent symbols, can accurately obtain the noise estimation value of the received signal, and further ingeniously utilizes the channel amplitude response value to reflect the relative power size of the received signal sequence. Finally, the fine-grained real-time LLR weight value can be obtained through the ratio of the two, so that the technical defects of the existing simplified model scheme, such as the large performance loss under complex channel conditions, and the fixed weight value scheme, such as the narrow adaptation range and the performance that cannot be fully released, are overcome, and the best decoding performance of the decoder is ensured.

The present application is suitable for the field of geophysical exploration technology, and provides a rock physics driven time-frequency domain P-wave velocity dispersion attribute prediction method, comprising the following steps: establishing a rock physics model; taking the rock physics model as the driving force, inverting the micro-fracture porosity in the well, calculating the skeleton elastic modulus of the rock containing pores and fractures, and predicting the frequency-dependent P-wave velocity logging curve; constructing a time-frequency domain P-wave velocity seismic inversion initial model; constructing a TF-AVO theoretical formula, developing a time-frequency domain P-wave velocity seismic inversion method based on the L-BFGS algorithm, and taking the initial model as the prior information to perform P-wave velocity inversion; analyzing the relative change of the frequency-dependent P-wave velocity under different frequencies, quantitatively predicting the P-wave velocity dispersion attribute; based on the logging data, calibrating the P-wave velocity dispersion attribute prediction result, and detecting the reservoir oil and gas. The present application has higher precision and resolution, significantly improves the stability and reliability of the P-wave velocity dispersion attribute prediction, and overcomes the dependence on strong amplitude response.

The invention discloses a variable fractional delay filter optimization method and system based on full coefficient compression. The method comprises the following steps: firstly, constructing a variable fractional delay filter frequency response function based on a Farrow structure; then setting an amplitude constraint based on a variable amplitude response peak error and a delay constraint based on a variable fractional delay peak error; setting constraint parameters based on a full-coefficient compression L0 norm, constructing an optimization model based on a full-coefficient compression sensing L0 norm, and converting the optimization model into an L1 norm optimization model; and finally, solving a weighted L1 norm optimization problem to obtain an optimal variable fractional delay filter coefficient. The system is used for realizing the variable fractional delay filter optimization method based on full coefficient compression. According to the method, the variable fractional delay filter with the optimal amplitude error performance can be obtained under the condition of least resource and time consumption, the amplitude error performance of the system is optimized, and the method is suitable for the field of radar signal processing.