1166 results about "Filter (signal processing)" patented technology

The present invention relates to the technical field of monitoring, in particular to a remote monitoring method and system for an aviation obstruction light. The method comprises the following steps: on the basis of an external sensor, acquiring electromagnetic signals sent by an aviation obstruction light; and by means of using a signal processing algorithm, processing the obtained original signals to eliminate noise interference and standardize the signal format, so as to generate signal-purified data. Using a support vector machine and a random forest algorithm in the present invention enhances the fault mode identification capability and the accuracy of predicting device performance degradation trends, and substantially improves the reliability of fault prediction; the combination of a Kalman filter and a multi-level decision tree provides powerful support for the integration and analysis of multi-source data, thereby ensuring the comprehensiveness and effectiveness of decision-making support information; and using a self-organizing map network and U matrix visualization technology not only shows advantages in the aspects of data mode identification and anomaly detection, but also improves the interpretability of data analysis by means of visual image displaying.

The invention discloses a power supply real-time compensation technology and application based on digital signal processing, and relates to the technical field of power electronics, and the power supply real-time compensation technology comprises a split-phase parallel processing architecture: a three-phase independently configured DSP core, each core integration comprises a real-time harmonic detection module, a dynamic sliding window FFT-RT algorithm is adopted, and the window length is adjustable in a set range; the bandwidth of the self-adaptive variable parameter filter can be dynamically adjusted in a set range; the FPGA coprocessor is specially used for PWM generation; an input stage of the harmonic prediction and decomposition module is synchronously acquired by a multi-source sensor; the processing layer comprises an improved VMD decomposition unit and a variational mode decomposition order; the edge prediction network is used for deploying a lightweight LSTM model, embedding a DSP core and predicting a harmonic spectrum in a future short period; and the multi-target dynamic optimization layer is used for setting a dynamic weight distributor and performing real-time adjustment according to the load sudden change rate. According to the invention, through split-phase parallel architecture, harmonic prediction and dynamic optimization, the response speed, multi-target cooperation and reliability improvement in the field of power supply real-time compensation are realized.

The invention discloses a calibration system in an active channel of a waveguide array phased-array antenna, and relates to the technical field of calibration in the active channel. Comprising a calibration judgment module, a waveguide directional coupling module, an amplitude-phase detection module and a digital domain optimization module, the calibration judgment module realizes intelligent triggering calibration through real-time monitoring of a multi-dimensional environment and performance parameters, and ensures that a calibration process is started only when the system state is deteriorated; the waveguide directional coupling module ensures that the calibration signal is accurately injected into each channel, and realizes cross polarization suppression through a polarization grid filter; the amplitude-phase detection module realizes high-precision measurement of amplitude, phase, group delay and polarization parameters of feedback signals of each channel by using superheterodyne and IQ demodulation technologies; and the digital domain optimization module calculates and updates the complex compensation weight of each channel through real-time digital signal processing and a self-adaptive algorithm, and realizes comprehensive and accurate calibration in combination with temperature-frequency interpolation and nonlinear pre-distortion compensation.

The invention belongs to the technical field of signal acquisition and processing, and discloses a low-noise biopotential signal acquisition system and a low-noise biopotential signal processing method. An original bioelectricity signal of a human body is collected through an electrode array, and analog-to-digital conversion is carried out after the signal is processed by a multi-stage self-adaptive filtering and amplifying circuit. A digital signal is sequentially subjected to adaptive wavelet transform denoising and empirical mode decomposition to obtain a multi-level intrinsic mode function set, an intrinsic mode function of a characteristic frequency band is extracted from the multi-level intrinsic mode function set, and the signal is reconstructed. Introducing a dual noise reduction mechanism combining adaptive wavelet transform and empirical mode decomposition; a signal quality real-time evaluation system is established, and system parameters including the gain of a variable gain amplifier and the bandwidth of a dynamic band-pass filter are dynamically adjusted through closed-loop feedback. The self-adaptive optimization of the whole biopotential signal acquisition process is realized, various interferences are effectively inhibited, the signal characteristics are reserved, and the signal-to-noise ratio is remarkably improved.

A system and method are disclosed for low-light image enhancement using hierarchical adaptive wavelet decomposition with cross-scale feature fusion. The system analyzes a raw input image to determine image characteristics and preprocessing parameters. A hierarchical adaptive wavelet decomposition process creates a variable-depth decomposition tree comprising frequency domain nodes, with decomposition depth determined by local image complexity. Cross-scale feature fusion implements attention mechanisms between nodes at different decomposition levels, enabling bidirectional information flow across scales. A dynamic network pool allocates specialized neural networks to process nodes based on their frequency characteristics, with weight sharing between similar nodes for efficiency. An adaptive reconstruction engine traverses the decomposition tree using learned filters and multi-scale residual learning to produce an enhanced image. The hierarchical approach enables superior low-light image enhancement by allocating computational resources based on content complexity, achieving better quality than fixed decomposition methods while maintaining compatibility with existing image signal processing pipelines.

An adaptive variable step-size LMS filter based on a hyperbolic tangent function and its filtering method and computer equipment belong to the field of filter technology, and solve the problem that the LMS algorithm improved based on a nonlinear function requires too many manual adjustments of parameters and that the convergence speed and convergence accuracy of the traditional LMS algorithm cannot be satisfied simultaneously. The method of the present invention includes: determining the filter tap length and the number of iterations; initializing the filter parameters; obtaining the filter input signal vector; determining the instantaneous expected signal of the filter; obtaining the instantaneous output signal; obtaining the instantaneous error signal of the filter and the accumulated amount of the error signal based on the instantaneous output signal; obtaining the instantaneous step size using a step-size update formula based on a hyperbolic tangent function; and continuously iterating the filter tap weight coefficient vector to construct an LMS filter using the filter tap weight coefficient vector obtained after the number of iterations. The present invention is suitable for improving the performance of the LMS filter, and the filter of the present invention can be widely used in various signal processing fields.

The invention discloses an automatic control method and system for a bias point of a photoelectric modulator, and relates to the field of signal processing, and the method comprises the steps: generating a composite disturbance voltage signal through a digital signal processor; the bias electrode is injected into the photoelectric modulator and used for collecting the optical signal output by the photoelectric modulator and converting the optical signal into an electric signal; extracting a fundamental component amplitude and phase difference data of a second harmonic component, a third harmonic component and a fundamental wave; dynamic characteristic parameters are calculated through spectral analysis, and the mapping relation between the characteristic parameters and the offset of the offset points is established; outputting the offset direction and the offset level of the offset point through a deep learning model; a bias voltage compensation amount is generated through a fuzzy PID (Proportion Integration Differentiation) controller; and adjusting the bias electrode voltage, smoothing the filter, and eliminating high-frequency jitter. The method has the advantages that high-precision and stable bias point adjustment is realized through the composite disturbance signal, real-time spectrum analysis and deep learning optimization decision in combination with fuzzy PID control and a dynamic closed-loop mechanism, and the system performance and stability are remarkably improved.

The invention relates to the technical field of particle accelerators, in particular to a transverse feedback system with a switchable excitation direction and a control method thereof, and the system comprises a beam position detector BPM which is used for measuring a beam signal; the RF front end is used for receiving the beam signal sent by the beam position detector BPM and carrying out amplification and filtering processing; the ADC module is used for converting the beam signal into a digital signal; the digital signal processing module is used for carrying out digital signal processing calculation to obtain an excitation signal; the DAC module is used for carrying out digital-to-analog conversion; the power divider is used for adjusting the phase of the excitation signal; the exciter comprises four Kickers with electrodes, a power amplifier, a low-pass filter and a matched load and is used for applying a transverse excitation signal to the beam current, and the problems that in the prior art, a transverse feedback system Kicker occupies a large longitudinal space, the system is complex, and precision and efficiency are low are solved.

The invention discloses a high-throughput satellite communication terminal system, which comprises a radio frequency channel unit, a signal processing unit, a terminal processing unit, a power supply management unit and an anti-electromagnetic interference module, and is characterized in that the radio frequency channel unit carries an agile device and a power amplifier and is used for realizing bidirectional conversion between a baseband signal and a radio frequency signal; communication frequencies of a Ka frequency band, a valley value of 5GHz and a peak value of 40GHz are supported, a zero intermediate frequency receiver structure is adopted, image frequency interference is eliminated through one-time frequency conversion to a baseband, and a high-integration design that a radio frequency signal directly enters a frequency mixer after passing through an antenna, a radio frequency filter and a low-noise amplifier is realized. According to the high-throughput satellite communication terminal system, the coverage problem of traditional communication in a blind area of a power grid is solved, safe, efficient and stable transmission of power data is realized, the emergency communication capability and the equipment environment adaptability are remarkably improved, and key technical support is provided for constructing global visual monitoring and intelligent scheduling of the digital power grid.

The invention belongs to the technical field of underwater acoustic engineering and digital signal processing, and particularly relates to a fractional delay FIR (Finite Impulse Response) phase calibration method and system of a multichannel sonar system. Comprising the steps of signal acquisition and reference channel selection; high-precision relative delay estimation; calculating a cross-correlation function of other channel signals and the reference channel signal, performing coarse estimation by searching a peak point of the amplitude of the cross-correlation function, and performing fine estimation on the peak position by using a parabola interpolation method to obtain the delay time of each channel relative to the reference channel; designing an FIR fractional delay filter; digital filtering and phase calibration; verifying a calibration effect; and calculating the phase of each channel signal at the target frequency point after calibration, and counting the standard deviation of each channel signal so as to evaluate the improvement degree of phase consistency. The method aims at overcoming the defect that traditional integral multiple sampling delay correction precision is insufficient, high-precision phase compensation of a sub-sampling point level is achieved, and therefore the performance of algorithms such as beam forming of a multi-channel sonar system is remarkably improved.

The invention relates to an ultralow-delay visual positioning system and method based on an event camera. The system comprises an event sensing module, an inertial measurement unit (IMU) and a mixed signal processing unit. The event sensing module has a microsecond-level response characteristic and is integrated with a dynamic bias adjustment and tunable optical filter; the IMU is rigidly connected with the IMU, and the physical center alignment error is less than 0.5 degree; and the hybrid processing unit adopts an FPGA (Field Programmable Gate Array) and ARM (Advanced RISC Machines) collaborative architecture to realize event filtering, time synchronization, motion compensation and tight coupling optimization. According to the method, an event brightness residual error and an IMU pre-integration residual error are combined to construct a joint cost function, a sliding window is dynamically adjusted based on event entropy, and loop-back and closed-loop correction is realized through mutual information criteria. The method can realize six-degree-of-freedom pose estimation as high as 500Hz, has the advantages of low delay, high precision, high robustness and the like, and is suitable for high-speed scenes such as unmanned aerial vehicles, AR navigation, industrial robots and the like.

The invention discloses an electroencephalogram signal processing method and system based on motion artifact prediction. The method comprises the following steps: acquiring experimental electroencephalogram data when a testee executes a motion imagination task; discrete wavelet transform is carried out on experimental electroencephalogram data, and signals are decomposed into low-frequency components and high-frequency components through a low-pass filter and a high-pass filter; inputting the low-frequency component into a pre-constructed and trained ARIMA model, and predicting to obtain a linear artifact; inputting the high-frequency component into a pre-constructed and trained XGBoost regression model, and predicting to obtain a nonlinear artifact; combining the linear artifacts and the nonlinear artifacts to generate a complete artifact prediction signal; real electroencephalogram signals are separated through difference value calculation of the experimental electroencephalogram data and the artifact prediction signals. According to the method, the time sequence change of the motion artifacts is predicted by utilizing the ARIMA model, and the nonlinear artifact features are captured in combination with the XGBoost regression model, so that the motion artifacts can be effectively removed, and purer electroencephalogram signals can be recovered.

The invention discloses a self-adaptive rotating baseline direction finding method, device and equipment, a medium and a terminal. Signals are continuously collected through the base line rotatable antenna array; rapidly identifying and outputting environment parameters by using the trained environment evaluation model, and evaluating the environment grade; constructing a dynamic direction-finding angle model by combining the attitude parameters and the motion parameters of the mobile platform, the control parameters of the antenna array and the environmental parameters; according to the environment parameters and the environment grades, a dynamic parameter optimization rule is formulated, and control parameters, direction-finding resolving parameters and signal processing algorithm parameters of the antenna array are adjusted; and based on the environmental parameters, the adjusted control parameters of the antenna array, the adjusted direction-finding resolving parameters and the adjusted adaptive filter parameters, calculating an incoming wave angle by using a dynamic direction-finding angle model. According to the method, high-precision incoming wave signal direction finding is realized in a dynamic environment, and the method is suitable for mobile scenes such as unmanned aerial vehicles and vehicle-mounted communication.

The invention belongs to the technical field of intelligent wireless sensing and radar signal processing, and particularly relates to a millimeter wave radar gesture recognition feature processing method, which is particularly suitable for scenes with environment interference (such as walking of others and static clutter), and specifically comprises the following steps: S1, preliminary filtering by a self-adaptive filter; s2, carrying out improved I CEEMDAN decomposition; s3, I MF component classification and processing; s4, signal reconstruction; the experimental result shows that the method provides a robust and efficient solution for gesture recognition, and can be widely applied to the fields of man-machine interaction, virtual reality, intelligent equipment control and the like.

The invention provides a full-parallel arbitrary multiplying power variable sampling system, and belongs to the technical field of signal processing. The system comprises an up-sampling system and a down-sampling system, the up-sampling system is used for a transmitter and comprises an up-sampling filter and an up-sampling interpolator, and the down-sampling system is used for a receiver and comprises a down-sampling filter, an extractor and a down-sampling interpolator; the up-sampling system can process four paths of input signals in parallel, and four paths of output signals are obtained after the four paths of input signals are processed in parallel; the down-sampling system is formed by cascading a plurality of down-sampling filters and an extractor, an extraction module is responsible for down-sampling of integer 2n times, and a decimal interpolator is connected behind the extractor to realize down-sampling processing of any decimal times. According to the invention, the limitation of traditional integer or simple fractional multiple sampling is broken through, arbitrary-multiplying-power precise resampling of the GHz-level signal is realized, the processing speed and the interpolation precision are obviously improved, the error is low, the anti-aliasing capability is strong, and the method is suitable for the high-requirement fields such as private network communication.

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

The invention relates to the technical field of fuel gas ultrasonic metering, and discloses a self-adaptive filtering method based on ultrasonic metering and ultrasonic metering equipment, and the self-adaptive filtering method comprises the steps: delaying an uplink echo signal for preset time to generate an uplink echo reference signal, and delaying a downlink echo signal for preset time to generate a downlink echo reference signal; respectively inputting the uplink echo signal and the downlink echo signal into a self-adaptive filter for filtering, outputting an uplink echo output signal and a downlink echo output signal, and performing iterative adjustment on a weight coefficient of the self-adaptive filter by using a variable step size, the minimum mean square value of the uplink echo error signal and the minimum mean square value of the downlink echo error signal meet the preset requirements. The self-adaptive model based on correlation is used for echo signal processing during ultrasonic metering instantaneous time difference acquisition, and the problem that interference is within a frequency sweeping range is solved.

The invention relates to the technical field of micro electro mechanical system sensors and signal processing, in particular to an acceleration sensor temperature drift compensation method and device and electronic equipment, and the method comprises the steps: obtaining the dynamic response data of an acceleration sensor at a plurality of working temperatures; fitting the dynamic response data to obtain a physical parameter and a continuous time transfer function of the acceleration sensor, and determining a digital filter coefficient according to the physical parameter and the continuous time transfer function; fitting based on the digital filter coefficient and the temperature to obtain a linear drift coefficient of the digital filter coefficient and the temperature, constructing a temperature offset compensation model according to the linear drift coefficient, inputting the working temperature of the acceleration sensor into the temperature offset compensation model, and outputting the digital filter coefficient by the compensation model, and performing temperature drift compensation on the acceleration sensor by using the digital filter coefficient. Therefore, the problems of output distortion caused by the dynamic characteristic of the acceleration sensor along with temperature drift, difficulty in realizing effective temperature self-adaptive dynamic compensation and the like are solved.

The invention discloses a weak radar signal target tracking method, system and device and a medium, and belongs to the field of radar signal processing, and the method comprises the steps: constructing a state space model, predicting a target state and an error covariance at a moment k based on model parameters and a target state filtering value and an error covariance filtering value at a moment k-1, and obtaining a target state and an error covariance; target motion information is obtained according to the echo signals at the moment k, whether dead pixels or leakage points exist or not is judged, if yes, the predicted position value in the target state at the moment k is extracted to replace the original position value, the target position error value and the mean square error are calculated again and replace the original error value and the mean square error, and the target position is obtained. Updating the filtering value into the predicted target state and error covariance at the moment k, and if not, calculating the Kalman gain and combining with the position value at the moment k, and performing filtering correction on the predicted target state and error covariance at the moment k; the system, the equipment and the medium are used for implementing the method. According to the method, the robustness and the tracking precision are improved, and the calculation complexity and calculation resource occupation are reduced.

The invention relates to a time domain broadband adaptive beam forming method and system based on Chebyshev weighting, and belongs to the field of array signal processing. Aiming at the technical problems of poor real-time performance and high sidelobe of the existing broadband beam forming, the method comprises the following steps of: dividing a received broadband signal into a plurality of sub-bands, obtaining an optimal weight vector for each sub-band by adopting a minimum linear constraint iterative algorithm after down-conversion, constructing an expected response of an FIR filter according to each weight, obtaining a filter coefficient through L2 norm convex optimization, and finally obtaining a high sidelobe beam according to the expected response of the FIR filter. And performing time domain weighting on each array element signal by using a filter bank, and outputting a directional beam. According to the method, the calculated amount is low, a main lobe and interference null can be formed in real time in a broadband, and a certain side lobe height is kept.

The present application relates to the technical field of speech processing, and particularly relates to a mixed speech processing method, an electronic device and a computer readable medium. The method comprises the following steps: collecting mixed speech and environmental influence parameters; performing bionics frequency domain analysis on the mixed speech to obtain low-frequency attenuation compensation feature data; performing multipath effect propagation analysis on the low-frequency attenuation compensation feature data through the environmental influence parameters to generate channel distortion data; performing time domain-frequency domain joint deconvolution processing on the mixed speech by using the channel distortion data to generate direct sound components and reflected sound components; performing adversarial training based on the direct sound components and the reflected sound components to generate anti-multipath speech enhancement data; and constructing a dynamic frequency compensation filter based on preset environmental acoustic characteristics. The present application improves the output quality of mixed speech through multi-stage signal processing, frequency compensation and real-time optimization technology.

The invention discloses a digital stethoscope system based on a piezoelectric film, and relates to the technical field of medical electronics and biomedical engineering. An existing digital stethoscope based on a piezoelectric film has the challenges of signal noise and interference, limitation of a signal processing algorithm, nonlinear response and time migration, insufficient diagnosis accuracy and automation and the like. A piezoelectric film and microphone dual audio acquisition mode is fused, various physiological signals such as low-frequency vibration and high-quality heart and lung sound are captured, and background noise is removed through a self-adaptive filter and a frequency spectrum subtraction technology; an MFCC feature extraction and dynamic time warping algorithm is utilized to carry out fine feature analysis and time axis alignment on the preprocessed audio signal, and high-precision matching with a storage template is realized; different physiological states are accurately distinguished through intelligent feature processing, real-time and visual diagnosis feedback is achieved through wireless transmission and state indication, and the reliability and efficiency of medical detection are improved.

The invention relates to the technical field of ultrasonic scanning microscopes, in particular to a system and a method for measuring surface unevenness and warping of an ultrasonic scanning microscope. The system comprises a water tank, a light emitting device, a light receiving device and a signal processing module, the light emitting device generates line laser through a semiconductor laser and an optical shaping module, and scanning is achieved through a high-precision galvanometer. The light receiving device is provided with a narrow-band filter, a telecentric lens and a CMOS sensor and is used for capturing a light spot image; the signal processing module integrates sub-pixel positioning, galvanometer nonlinear correction, self-adaptive filtering and temperature compensation algorithms, light spot center sub-pixel-level positioning is achieved through Gaussian curved surface fitting, galvanometer correction and temperature drift compensation are combined, the measurement precision and stability are improved, and finally high-precision height information is fed back to an ultrasonic scanning system. According to the invention, the accuracy and reliability of surface unevenness and warping measurement are improved.

The invention relates to the field of radar detection and signal processing, and particularly discloses a radar data technology system for monitoring a tiny target, and the system comprises a signal receiving and transmitting module which is used for generating and transmitting a linear frequency modulation signal based on a linear frequency modulation technology, and receiving an original echo signal reflected by the target; the pulse compression processing module is used for performing complex convolution operation on the original echo signal based on a matched filter to generate a compressed pulse signal; the multi-pulse coherent accumulation module is used for performing frequency domain energy accumulation on a plurality of compressed pulse signals of the same distance unit based on a discrete Fourier transform technology; and the Kalman filtering tracking module is used for processing the signals after multi-pulse coherent accumulation through a Kalman filtering algorithm, predicting and updating a target state in real time, and generating final trace point data. According to the invention, through key technologies of large-time wide-bandwidth pulse compression, multi-pulse coherent accumulation and the like, the detection capability of the radar on a tiny target is remarkably improved.

This disclosure provides methods, devices, and systems for audio signal processing. The present implementations more specifically relate to speech enhancement techniques for separating microphone signals into speech, echo, and noise signals. In some aspects, a speech enhancement system may include a delay estimator and an acoustic echo and noise (AEN) decoupling filter. The delay estimator receives a microphone signal via a microphone and a far-end audio signal for output via a speaker and estimates a reference audio signal based on a delay between the microphone signal and the far-end audio signal. In some aspects, the AEN decoupling filter may determine a speech mask, an echo mask, and a noise mask based on the microphone signal and the reference audio signal and may suppress an echo component and a noise component of the microphone signal based on the determined set of masks.

An apparatus for calibrating a digital filter to replicate a transfer function of a signal processing device comprising a detector with a first input to receive a first signal; a second input to receive a response signal of the signal processing device to the first signal; a signal modification block; a comparison block and a decimation block; wherein the comparison-block compares a phase and amplitude of the first signal after a correction has been applied by the signal modification block and decimation has been applied by the decimation block; and a feedback loop; wherein detector is configured to determine at least a feedback control signal at a first frequency and a second frequency, different to the first frequency and determine calibration information for programming of the transfer function of said digital filter.

The invention belongs to the technical field of nondestructive detection defect evaluation, and particularly relates to an electromagnetic regulation and control detection method and system for structural defects of offshore oil well control equipment. According to the electromagnetic regulation and control detection method and detection system for the structural defects of the offshore oil well control equipment, crack multi-feature signals are obtained through amplitude-frequency phase regulation and control of electromagnetic excitation signals, angle information of cracks is obtained by means of a residual network, and accurate identification of crack length and depth information is achieved; and the service life prediction network realizes effective prediction of the residual fatigue life. The detection system comprises a monitoring probe, an excitation signal generation unit and a monitoring signal processing and collecting unit, the monitoring probe consists of an exciting coil group, an array sensor and a front-end signal processing circuit; the excitation signal generation unit comprises a signal generator and a power amplifier; the monitoring signal processing and collecting unit comprises a multiplexing and amplifying filter circuit, a band-pass filter circuit, a phase-locked amplifying circuit and a signal collecting board card.

The invention particularly relates to a voice noise reduction method suitable for different noise environments, and relates to the technical field of voice signal processing. Analyzing a noise environment and characteristics; adaptive selection of a noise reduction algorithm; adaptive noise reduction processing is executed; and noise reduction effect evaluation and iterative optimization are carried out. According to the invention, refined noise classification is matched with the algorithm, so that the limitation of a traditional single algorithm in a complex environment is solved; the steady-state noise is dynamically counteracted by adopting an adaptive filter, the frequency domain gain of the unsteady-state noise is adjusted in real time through Wiener filtering, and the pulse noise is subjected to dynamic threshold processing by combining median filtering and wavelet transform; especially, the design of cooperation factors in wavelet transform can accurately distinguish signal details and noise: intensively suppress pulse noise, loosely retain details such as voices and consonants, and realize dynamic balance of noise reduction intensity and signal distortion.

The invention relates to the technical field of radio frequency signal processing, and discloses a digital filtering repair method and system for signal distortion, and the method comprises the steps: obtaining a frequency domain amplitude-frequency response and a phase response; constructing a phase compensation amount by adopting piecewise polynomial fitting and a phase inversion mechanism; a complex number logarithm domain mapping mechanism is adopted to execute composite frequency response inversion; constructing a switchable hybrid digital filter with an amplitude path and a phase path coupled in parallel; multi-band adaptive scheduling is executed; and outputting the radio frequency baseband signal stream. The technical problems that in the prior art, frequency response nonlinear distortion in a broadband range cannot be dealt with by adopting a fixed template or a single-path filter, and particularly, a high-precision and low-delay dynamic compensation effect cannot be realized in a scene that multi-band coupling interference exists in a radio frequency ADC channel with the bandwidth exceeding 800MHz are solved. Due to the fact that the switchable hybrid digital filter is constructed, accurate modeling and flexible recovery of distortion characteristics in broadband signals are achieved.

The invention discloses a filtering adjustment method and system based on an adaptive clock, and relates to the technical field of filtering adjustment, and the method comprises the steps: carrying out the time domain framing processing of a collected input signal through employing a sliding window, and combining DCT and compressed sensing to reconstruct the signal, and the main frequency of the signal and the dynamic change of the main frequency in the time sequence are accurately extracted through FFT, Gaussian pyramid layering and a multi-scale optical flow method, so that the frequency identification capability of the non-stationary signal is effectively enhanced, the frequency spectrum resolution is ensured, the data processing amount is remarkably reduced, the response speed of the signal frequency change is improved, and the method is suitable for popularization and application. Through the main-sub clock structure, the quick response of the center frequency of the filter to the change of the main frequency is ensured, the selectivity of the filter and the signal fidelity are improved, through the compensation adjustment of environmental factors, the influence of temperature and voltage changes on signal processing can be effectively reduced, and the robustness and stability of the system are enhanced.