373 results about "Oversampling" patented technology

The invention discloses a sampling implementation method and system suitable for motor protection quick response, belongs to the technical field of motor protection, and can increase the sampling frequency and improve the quick response of motor fault protection. Comprising the steps that motor phase current and bus voltage are collected, after anti-aliasing filtering is conducted, parallel sampling is conducted through a main ADC module and a redundant ADC module which are independently configured, and the following operations are synchronously executed based on a double-buffer-area framework: in a first buffer area, a compensation coefficient updating period is dynamically adjusted, and convergence is restrained; in the second buffer area, generating a voltage gradient prediction sequence, and compensating data abrupt change; when the harmonic frequency spectrum amplitude or the voltage gradient prediction sequence exceeds the limit, an oversampling mode is triggered, a redundant ADC module is allocated to load a multi-order digital filter, and an independent hardware acceleration unit is started to process harmonic compensation and gradient prediction in parallel; and generating a dynamic safety envelope, and if the gradient extreme value in the preset continuous window breaks through the safety envelope, generating an overvoltage fault signal.

An electronic interface device is configured for use with associated optics for employing coded light for illuminating a liquid, gaseous, or other analyte or target, such as for measuring or determining a composition or other property of the analyte or target. The arrangement can help enable a far higher level of performance, such as for spectral analysis, imaging, or both. A shared common clock on both the illumination and receive sides of the system can be included, such as together with one or more of reference detection, per-channel wavelength-dependent modulation, multiple concurrent feedback loops at different levels, orthogonal and circulant codes, phase-scrambling to better utilize ADC dynamic range, oversampling, and other signal processing can be provided.

Disclosed in the present invention are an improved oversampling-based power transmission line fault classification system and method. The method of the present invention comprises: first, by using historical fault information of State Grid as a database, obtaining a plurality of sub-datasets by means of division based on fault types, each sub-dataset consisting of transient current traveling waves at power transmission line fault moments; then, using a K-means method to perform clustering operation on minority classes of a fault dataset, and using BSMOTE to perform oversampling, so as to achieve balancing of various classes, thus forming a balanced dataset; then, calculating time domain and frequency domain features of transient current traveling waves in the balanced dataset, so as to form a feature dataset; finally, on the basis of CNN and SVM deep learning algorithms, establishing a classification model and training same by using the feature dataset, so as to obtain a trained fault classification model; and using the trained fault classification model to classify and identify faults. The method is highly accurate and effective, and can effectively extract fault feature quantities and quickly and accurately classify faults.

The invention provides a VCO digital circuit control method and system with temperature compensation, and relates to the technical field of circuit control, and the method comprises the steps: obtaining VCO output frequency and real-time temperature, and calculating an initial frequency deviation; correcting the frequency deviation based on the temperature compensation coefficient, and generating a temperature-frequency deviation characteristic curve; an optimal compensation factor is determined through Monte Carlo sampling; calculating a compensation parameter adaptive weight by using Bayesian iteration, and dynamically adjusting a compensation factor; and a control signal is generated through the digital integrator to realize frequency calibration. According to the invention, the high-precision frequency stability of the VCO under the temperature change condition is realized, and the circuit reliability is improved.

The invention discloses a data transmission method, a server system, equipment and a medium, and relates to the technical field of data transmission, and the method comprises the steps: determining a sampling pulse corresponding to first target data, obtaining a plurality of sampling pulses in a segment through the segmentation processing of the sampling pulse, and taking the plurality of sampling pulses as target sampling pulses, so as to improve the accuracy of collecting effective data. According to the method, interference signals caused by corresponding jitter instability conditions under serial port hot plugging are avoided, and the target sampling pulses obtained after segmentation processing are used for keeping data stability and avoiding high possibility of doping of the interference signals at the same time. And determining the second target data based on the target sampling pulse, and improving the anti-interference performance of the second target data under the bit. The technical problems that normal serial port use cannot be maintained when jitter interference occurs in the hot plug process, and even the whole machine crashes can be solved, and the technical effect of improving the anti-interference performance of data is achieved.

The invention belongs to the technical field of cognitive radio, and particularly relates to an open set modulation identification method based on time-frequency domain feature learning and fusion. The method comprises the following steps of: receiving a signal by adopting a receiving antenna through electromagnetic spectrum monitoring equipment, capturing a space electromagnetic radiation signal, converting the space electromagnetic radiation signal into an electric signal, amplifying the signal through a low-noise amplifier, performing frequency conversion and filtering through an analog receiver, and converting the received signal into an intermediate-frequency signal. Then, the signal is digitalized through AD sampling, and the intermediate frequency signal is converted into a baseband complex signal through frequency mixing and digital filtering in a digital receiver. Then performing time domain characterization calculation and frequency domain characterization calculation on the signal sampling data to form a plurality of time domain characterization vectors and a plurality of frequency domain characterization vectors; and then the vectors are sent to a trained deep neural network model, reasoning is carried out by using the network model, a reasoning result is post-processed, and a signal modulation identification result is finally obtained. According to the invention, the recognition accuracy of unknown signals is improved.

The invention discloses a low-complexity MSK coded modulation system phase synchronization receiving system and method, and relates to the technical field of continuous phase modulation. On one hand, an original message is coded and modulated by constructing a pre-coding mode, a receiving end realizes optimal sampling point extraction at the minimum intersymbol crosstalk moment after phase decoupling frequency conversion processing, MSK is converted into BPSK-like modulation, oversampling processing is not needed, modulation phase components of a receiving synchronization section can be offset, and the modulation efficiency of the receiving synchronization section is improved; therefore, accurate estimation of Doppler phase and frequency is realized, and the anti-Doppler capability is improved. On the other hand, coherent demodulation soft information is directly output after the real part of the signal subjected to Doppler frequency offset and phase offset compensation is obtained, the complexity of the demodulation algorithm is low, the performance of the demodulation algorithm is equivalent to that of BPSK, the demodulation algorithm is remarkably superior to that of an MSK optimal demodulation receiver in a low signal-to-noise ratio region, and the demodulation algorithm is particularly suitable for a bit interleaving coding modulation system combined with channel coding and has a wide application prospect. And reliable transmission under a low signal-to-noise ratio can be realized.

The invention discloses a fast-locking low-jitter sub-sampling phase-locked loop structure. The sub-sampling phase-locked loop comprises a sub-sampling Bang-Bang phase discrimination circuit, a self-adaptive variable step size digital filter circuit, a delta sigma digital-to-analog conversion circuit and an oscillator circuit. The sub-sampling Bang-Bang phase discrimination circuit generates a phase discrimination output digital control signal according to an input reference signal and a feedback oscillator signal phase error, the adaptive variable step size digital filter circuit adaptively adjusts a filtering gain according to the phase discrimination output digital control signal, and the output of the filter circuit realizes digital-analog conversion after oversampling modulation. And the output analog signal controls the voltage-controlled oscillator to realize rapid low-jitter phase locking of the output and the reference signal.

The invention relates to the technical field of digital playing equipment, and provides a lossless control method for the volume of a digital player and the digital player. The method comprises the following steps of: after receiving a target volume parameter, executing dynamic multiplying power oversampling and high-order noise shaping on an input digital audio signal, and injecting preset jitter to suppress in-band quantization distortion so as to generate an optimized digital audio stream; meanwhile, a high-resolution digital reference current control instruction corresponding to the target volume is independently generated, output of a reference current source is adjusted according to the high-resolution digital reference current control instruction, and a high-precision differential current signal is obtained by combining nonlinearity, temperature excursion and quantization errors of a closed-loop feedback compensation device through a multi-bit current segmentation network; performing bias stabilization processing on the signal, and converting the signal into a differential voltage signal for suppressing common-mode noise through a precision differential resistance network; and finally, low-distortion differential analog audio signals capable of directly driving a load are output through low-distortion gain, impedance conversion and power driving, so that low-distortion, high-signal-to-noise-ratio and wide-dynamic-range output in a full-volume range is realized.

The invention relates to a current-to-voltage acquisition circuit for suppressing common-mode interference. The current-to-voltage acquisition circuit comprises a current amplification module, a current transformer, a high-pass filter, an interference acquisition module, an interference analysis module and a suppression execution module, by means of the design, the common-mode interference suppression effect is better than that of a traditional sampling circuit, errors caused by sampling signal deviation generated by interference are avoided to the maximum degree, the common-mode interference suppression circuit is particularly suitable for interference scenes or occasions where interference sources change, the changed interference sources can be calculated in a sampling statistics mode, and the interference source suppression effect is improved. Therefore, the suppression ratio of common-mode suppression is adjusted, and the environment condition is dynamically tracked to obtain an accurate sampling effect.

The invention relates to the technical field of pulse acquisition, in particular to a high-speed ADC (Analog to Digital Converter) and FPGA (Field Programmable Gate Array) coordinated extremely-narrow pulse signal acquisition system, which comprises a phase identification monitoring module, a phase lock reference updating module, a sampling permission control module, a prohibition feedback identification module and a synchronous trigger adjustment module. According to the method, a beat tolerance judgment mechanism is introduced, hyper-bound phase deviation trend statistics and frequency vector extraction are combined, beat correction is carried out by utilizing a directional hopping cumulant and a period adjustment amount, a sampling permission judgment and keeping window alignment mechanism is introduced in a multi-frame period, and a two-way judgment means of a sampling starting trigger value is adopted, so that the beat tolerance of the multi-frame period is judged. And meanwhile, centralized extraction and frequency directivity statistics are carried out on the hopping direction in the sampling prohibition time sequence, the synchronous mismatch trend is distinguished, and the initial beat position is adjusted, so that the time domain characteristics of the narrow pulse waveform are kept continuous, the edge response is prevented from deviation, the trigger structure is stable and reliable, and the overall sampling judgment is more accurate and efficient.

The invention relates to the technical field of flow metering, in particular to an anti-flow-fluctuation metering method of a multichannel ultrasonic flowmeter. The method comprises the following steps: dividing the measurement of each sound channel into a plurality of measurement periods, calculating the weight proportion of each sound channel in the overall measurement based on a multi-factor weighting model, and introducing a real-time compensation mechanism of a fluid dynamic working condition to correct the weight proportion; when the fluid passes through the measuring pipe section, sound wave transmitting and receiving signals are sampled, the transmission time of sound waves in the fluid is calculated through sampled data, and the effective transmission time of each sound channel is recorded; and judging the effectiveness of sound channel sampling based on the sampled data and the effective transmission time of the sound channel, and establishing a multi-sound channel weighted average flow rate model. According to the invention, by introducing a multi-factor weighting model and a fluid dynamic working condition real-time compensation mechanism, the weight in the whole measurement can be dynamically adjusted according to the signal quality of each sound channel.

A circuit and method for data recovery comprises clock generation, data reception, data oversampling, and data selection circuits; wherein, clock generation circuits are used to output data reception clock signals and data processing clock signals; data reception circuits are used to receive original transmission data from the data transmitter in accordance with the data reception clock signals, and transmit data based on its output; data oversampling circuits are used for multiple oversampling of at least one bit of transmission data corresponding to at least one data processing cycle according to the data processing clock signal and outputting corresponding parallel sample data; data selection circuits are used for transition detection on the corresponding parallel sample data for each data processing cycle and selecting sample data from the corresponding parallel sample data to output based on the transition detection results. The embodiments of the present application use a simple structure to implement data recovery, reducing layout area and circuit power consumption.

The invention discloses a power quality composite disturbance detection classification and time positioning method based on MS-TCN + +. According to the method, composite disturbance category codes are designed for sampling points, an electric energy quality identification model comprising a prediction generation stage and a plurality of refining stages is constructed on the basis of MS-TCN + +, and multi-time-scale disturbance signal features are extracted by using double expansion layers in the prediction generation stage. And smooth loss is added in model training to avoid sudden change of categories of sampling points in a time sequence, and finally category identification and time positioning of disturbance signals are completed through a classification result of the sampling points, so that powerful support is provided for treatment of power quality disturbance.

A method for motor voltage reconstruction by Analog to Digital Converter (ADC) oversampling and averaging includes measuring a first phase voltage with an ADC, the first phase voltage proportional to a first duty cycle of a first Pulse Width Modulation (PWM) signal. A second phase voltage is measured with the ADC, the second phase voltage proportional to a second duty cycle of a second PWM signal, wherein the first PWM signal and the second PWM signal control consecutive phases of a three-phase motor. A third duty cycle of a third PWM signal of the three-phase motor is reconstructing from the first phase voltage and the second phase voltage.

The invention discloses a multi-scale epileptic seizure cross-object detection method based on power spectral density. The method comprises the following steps: acquiring multi-channel electroencephalogram signal data of an epileptic and preprocessing the multi-channel electroencephalogram signal data; carrying out regularization processing on the preprocessed electroencephalogram signal data; constructing self-adaptive power spectrum density; according to the power spectral density characteristics, balancing the electroencephalogram signal data by using an SMOTE oversampling method to form a balanced data set; building and training a multi-scale space-time deep convolutional network model; dividing data in the balanced data set into a training set and a verification set, and inputting the training data and the verification data into the multi-scale space-time deep convolutional network model for training. According to the method, the dependence on a large amount of marked data can be reduced, the complexity and the calculation cost of the model are reduced, and meanwhile, the applicability of the model among different patients is improved.

The invention relates to the technical field of electromagnetic valve internal resistance adjustment, in particular to an electromagnetic valve state monitoring device and method.The electromagnetic valve state monitoring device comprises a controller, an RC filter circuit, a sliding filter circuit, a drive circuit and an ADC sampling circuit, and the controller is connected with the RC filter circuit, the drive circuit and the ADC sampling circuit; the ADC sampling circuit is respectively connected with the driving circuit and the sliding filter circuit; wherein the sliding filter circuit maintains three historical data thresholds to form a sliding window, new sampling data is compared with each historical value in the window, sampling noise is eliminated, then the new sampling data is transmitted to the controller through the ADC sampling circuit, the controller collects the sampling data through the ADC sampling circuit and carries out data preprocessing, and therefore the on-off state of the electromagnetic valve is judged; according to the invention, the transient response characteristics of the driving voltage of the electromagnetic valve are fully utilized, the dynamic model solution and the adaptive filtering algorithm are matched, the effective signal reflecting the internal resistance change is separated, and high-precision state recognition and fault early warning are realized.

The invention discloses an anti-high-frequency bias sampling device and method. The sampling position of the next beat is updated in real time by comprehensively considering the hopping condition and the sampling position of the data to be oversampled of the previous beat, the data to be oversampled of the current beat is sampled based on the latest sampling position, and finally the sampled data with the dynamic bit width is output with the fixed bit width. Compared with the prior art, the method has the advantages that higher frequency offset can be tolerated by updating the sampling position in real time; the logic is simple, and does not need to calculate source and sink end frequency offset like a traditional method, so that more resources can be saved; the method is high in universality, and can be applied to bearer networks such as Ethernet, optical transport networks and the like, and even all places needing oversampling in digital communication.

The invention provides a communication radiation source individual identification method based on time domain and time sequence coding constellation diagram feature fusion. The method is used for realizing stable identification of different transmitters under non-ideal conditions of carrier frequency offset, noise interference and the like. The method comprises the following steps: firstly, carrying out normalization and fixed-length interception on a received I / Q signal, and calculating an oversampling multiple according to a symbol rate and a time resolution; then, a time domain and time sequence coding constellation diagram double-branch feature extraction network is constructed, a time domain structure feature is extracted from a time domain branch by adopting a one-dimensional modified ResNet18, and a constellation diagram with time information is generated from a time sequence coding constellation diagram branch through a time sequence coding constellation module and the feature that the phase changes along with time is extracted; and carrying out joint modeling on the two types of features through a trainable weighted fusion mechanism. In the training stage, the performance of the model is improved by adopting cross entropy loss, an Adam optimizer, learning rate scheduling and an early stop strategy. In the reasoning process, input signals are classified by using the model weight with the highest verification accuracy, and a communication radiation source individual identification result is output. Experimental results show that the method can effectively enhance the frequency offset robustness and generalization ability of the model, realizes high-precision identification of the individual communication radiation source, and is suitable for the fields of security access of the Internet of Things, spectrum monitoring, electronic countermeasures and the like.

The invention relates to the technical field of broadband signal synthesis and high-speed digital-to-analog conversion, in particular to an oversampling-based real-time frequency interleaving digital-to-analog conversion system aliasing suppression method and system, and aims to solve the problem of spectrum aliasing caused by non-ideal characteristics of a digital sub-band decomposition filter and an analog filter under a real-time architecture of the system. In the invention, the sub-band decomposition is completed by adopting the linear phase complementary filter bank, so that distortion introduced by a transition band of a digital filter is avoided; by restraining system parameters such as passband and stopband frequencies and local oscillation frequency of a sub-band decomposition filter, a protection band is set by an oversampling mechanism, a frequency margin is reserved for a transition band of the filter, and aliasing components are suppressed. According to the method, extraction aliasing of a digital domain and mirror image aliasing of an analog domain are avoided at the same time from the system architecture, the frequency spectrum purity of output signals is improved, the method is suitable for a real-time system for processing continuous data streams, and technical support is provided for achieving a high-performance broadband signal source.

An electronic interface device is configured for use with associated optics for employing coded light for illuminating a liquid, gaseous, or other analyte or target, such as for measuring or determining a composition or other property of the analyte or target. The arrangement can help enable a far higher level of performance, such as for spectral analysis, imaging, or both. A shared common clock on both the illumination and receive sides of the system can be included, such as together with one or more of reference detection, per-channel wavelength-dependent modulation, multiple concurrent feedback loops at different levels, orthogonal and circulant codes, phase-scrambling to better utilize ADC dynamic range, oversampling, and other signal processing can be provided. Modulation codes can optionally employ a clock divider, such as which can be configured to satisfy one or more specified constraints.

The invention provides a multiplexed signal parallel demultiplexing method based on an FPGA and a demultiplexing equalizer. The method comprises the following steps: receiving multiple paths of serial multiplexed signals from a high-speed link, performing oversampling processing on the received multiple paths of serial multiplexed signals, and performing parallelization processing on each path of serial multiplexed signals subjected to oversampling processing; inputting each path of parallelized signal into a butterfly equalization demultiplexing module, performing equalization processing on the input by using a periodically updated tap coefficient of the butterfly equalization demultiplexing module, and performing demultiplexing processing by using a CMA algorithm to obtain a received output signal for demultiplexing a multiplexing signal; sequentially selecting output signals of the butterfly equalization demultiplexing module at intervals according to a preset proportion, and calculating a tap coefficient accumulative error vector by combining the selected output signals and the corresponding input signals; and according to the iteration step length of the CMA algorithm, the tap coefficient of the butterfly equalization demultiplexing module is updated based on the tap coefficient accumulative error vector.

The invention discloses an OFDM (Orthogonal Frequency Division Multiplexing) signal time-frequency collaborative equalization distortion suppression method, and belongs to the field of wireless communication. According to the invention, the problems of high complexity and poor transmission reliability of the existing short cyclic prefix method are solved. According to the invention, an OFDM system based on time domain oversampling and frequency domain interpolation equalization is adopted at a receiving end, and the oversampling algorithm can enhance the power of useful signals by properly broadening an equalization frequency band, so that the equalization effect is enhanced; according to the frequency domain interpolation equalization algorithm, weighted interpolation processing is carried out on frequency domain receiving signals, so that spectrum details can be better captured, and compensation and correction of an energy migration phenomenon generated by spectral lines in a frequency band due to a channel effect are realized. Through time-frequency cooperative processing at a receiving end, interference can be efficiently suppressed, stable and reliable demodulation precision and communication quality can be ensured, frequency spectrum resource loss caused by additional extension of a cyclic prefix can be avoided, and the frequency spectrum utilization efficiency of the system is further remarkably improved. The method can be applied to the field of wireless communication.

The invention discloses a loudspeaker detection system in a high-reliability demand environment, and relates to the technical field of loudspeaker detection, and the loudspeaker detection system is used for generating and outputting a preset level signal, and supporting the switching of a single-path manual detection mode, a multi-path manual detection mode, an automatic multi-path detection mode and an aging detection mode; output signals are collected in real time through a current sampling resistor connected to the output end of the loudspeaker in series, sound frequency and intensity parameters of the output signals are analyzed, and whether the loudspeaker function is normal or not is judged; wherein the frequency parameter is consistent with the output signal frequency of the driving board, and the intensity parameter is positively correlated with the output signal current; sampling results of multi-channel manual detection are received, cross interference data of multi-channel test units are collected at the same time through a sampling resistor, and a compatibility rating result is generated; and receiving data of automatic multi-path detection as a health index calculation benchmark, and evaluating a performance attenuation trend and residual life.

Methods, systems, and devices for wireless communications are described. A user equipment (UE) may transmit uplink signaling via a cyclic prefix orthogonal frequency division multiplexing (CP-OFDM) waveform or a discrete Fourier transform (DFT) spread OFDM (DFT-s-OFDM) waveform. A configured resource muting pattern may be based on or may reuse a phase tracking reference signal (PTRS) frequency pattern. The resource muting pattern may mute every other resource element (RE) of an uplink symbol or uplink subband of subband full duplex (SBFD) symbols, and the transmitting UE may perform uplink rate matching (e.g., applying a zero power PTRS pattern and inserting the zero power REs after a DFT spread operation, oversampling after a DFT-s-OFDM procedure and doubling a size of an IFFT, or repeating an input sequence to a DFT and doubling a size of the IFFT, power boosting, updating a power control calculation, etc.).

The invention relates to the technical field of radars, in particular to a weather radar distance resolution calibration method and system. The method comprises the following steps: collecting oversampling original baseband I / Q data of a point target in a plurality of continuous pulse periods; determining a pulse signal-to-noise ratio, and screening out effective pulses according to a threshold value; carrying out sub-sampling stage time alignment on the effective pulses based on cross-correlation analysis, and carrying out incoherent averaging on power module values of all the aligned pulses to generate an average pulse power envelope; converting the average pulse power envelope to a logarithm domain, and selecting discrete data points in a main lobe region; according to the discrete data points, a nonlinear least square method is adopted to carry out iterative fitting on the generalized Gaussian mathematical model, and a continuous analytic function describing the pulse envelope is obtained; according to the continuous analytic function, determining a pulse time width when the power is reduced to a peak value of-6dB; and determining the real distance resolution according to the pulse time width. The measurement robustness is improved, and sub-meter calibration precision is achieved.

The invention discloses an aliasing suppression strategy based on oversampling filtering. The aliasing suppression strategy comprises a mixed time frequency analysis module, an oversampling processing module, a delta-sigma modulator module, an RC filter module and a Sinc filter module. Wherein the mixed time-frequency analysis is used for performing frequency band analysis on a signal and judging whether an aliasing phenomenon exists or not according to a result; the oversampling processing is used for carrying out high-resolution digital sampling on the signal so as to fully capture the periodic change of the signal; the delta-sigma modulator is used for converting the oversampled signal into a high-speed and low-resolution sequence; the RC filter is used for converting a high-speed 1-bit sequence into a signal in an analog form, removing high-frequency noise and smoothing the signal; the Sinc filter further improves the frequency characteristic of the signal and inhibits residual high-frequency noise, a cleaner and smoother analog signal is obtained, and the analog signal is more suitable for subsequent signal processing and analysis. The method has higher signal reconstruction precision, stronger anti-aliasing and anti-noise capabilities, and greater flexibility and adjustability, and is suitable for various application scenes.

The invention discloses a fault detection method for a flexible DC power transmission system and a related device, and the method employs a voltage gradient algorithm to detect whether the flexible DC power transmission system breaks down, and carries out the fault type discrimination based on the ground model voltage polarity through a sampled voltage signal when the system breaks down. And determining a high-frequency impedance difference value ratio of a rectification side and an inversion side in the flexible DC power transmission system based on the voltage signal and the sampled current signal, carrying out fault direction discrimination based on the high-frequency impedance difference value ratio, and finally determining a fault detection result according to a fault type and a fault direction. Therefore, the fault pole selection is realized based on the ground model voltage polarity, and the internal and external faults are accurately distinguished through the high-frequency impedance difference ratio, so that the reliability of fault detection of the flexible direct-current power transmission system is improved.

The invention relates to a dual-channel high-sensitivity signal spectrum monitoring method and system. The method comprises the following steps: acquiring two paths of signals synchronously acquired by two data acquisition devices, performing cross-correlation, and then calculating a cross-correlation module value; peak value searching is carried out according to the cross-correlation module value, oversampling analysis is carried out according to a peak value searching result to obtain an oversampling multiple, and a bandwidth estimation value is calculated according to the oversampling multiple and the acquisition rate of the data acquisition equipment; carrying out time delay compensation on the two paths of signals and then carrying out signal combination to obtain a combined signal; performing time-frequency conversion on the combined signal, and splicing data with a front effective window length in a time-frequency conversion result at the tail of the time-frequency conversion result to obtain reconstructed frequency domain data; and performing sliding correlation processing on the reconstructed frequency domain data according to the frequency domain filtering coefficient, performing peak detection according to a sliding correlation result to obtain a frequency point initial position, and calculating a center frequency point according to the bandwidth estimation value and the frequency point initial position. By adopting the method, the reliability of weak signal spectrum monitoring can be improved.

The utility model discloses a real-time vector detection device based on a radio frequency signal short-wave antenna tuner, which comprises a radio frequency signal sampling module, an AGC, an AD and an FPGA, the sampling input end of the radio frequency signal sampling module is connected with a radio frequency path, and the output end of the radio frequency signal sampling module is provided with a forward signal end and a reverse signal end and is connected to the AGC; the AGC can perform gain attenuation processing on an input forward signal and an input reverse signal, an output end of the AGC is provided with a forward signal end and a reverse signal end which are respectively connected to a sampling input end of the AD, the AD is a dual-channel over-sampling analog-to-digital conversion module, an output end of the AD is connected with the FPGA, an input clock of the FPGA is 100MHz, the DDS is realized by adopting an IP core design, the FPGA can realize sampling, calculation and frequency mixing, and the FPGA is connected with the FPGA. And the matching network is controlled to realize tuning and system protection or alarm according to the calculated impedance value and standing-wave ratio. According to the utility model, the hardware circuit is simplified, the power standing-wave ratio condition of the access can be detected in real time, the detection efficiency is improved, the use is convenient, and the detection is efficient and accurate.