1237 results about "Digital signal processing" patented technology

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 relates to the technical field of digital signal processing, in particular to a big-data-oriented digital signal processing method and system, and the method comprises the following steps: obtaining sampling data, recognizing an abnormal boundary, extracting polarity inversion frequency, analyzing synchronism, calculating a relative time difference to generate a deviation graph, and carrying out the clustering to obtain a signal synchronous reconstruction data set. According to the invention, fine positioning of the abnormal section boundary is realized through difference and extreme value determination, sensitivity and accuracy of signal boundary identification are enhanced, polarity inversion frequency statistics reveals a local fluctuation characteristic change trend, and expression ability of an internal disturbance structure of the abnormal section is improved. A synchronization index is constructed to reflect processing consistency and response synchronization characteristics among threads, relative time difference alignment analysis is performed to quantify the thread time sequence drift degree, the recognition depth of asynchronous behaviors is enhanced, and a density distribution clustering and gradient normalization mode reveals a local disturbance rule of sampling behaviors. And the processing stability and the detail identification capability of the big data in a dynamic environment are improved.

The invention relates to the technical field of encoders, in particular to an incremental optical encoder signal error compensation system and method.The incremental optical encoder signal error compensation system comprises a signal acquisition module, a preprocessing module, an error analysis module, a compensation operation module, a data storage module and an output control module; the preprocessing module preprocesses the original signal; the error analysis module analyzes and identifies the error type and characteristics of the processing signal in a multi-dimensional manner, and calculates compensation values of the scribing error, the subdivision error and the eccentric error through the compensation operation module; a compensation value is superposed with an original signal, a compensated high-precision signal is obtained, the output control module outputs the high-precision signal to a subsequent device, meanwhile, system parameters are adjusted according to an external instruction, a high-precision error model is constructed by the system, and the high-precision error model is obtained by combining an advanced digital signal processing technology and an intelligent algorithm. And the scribing error, the subdivision error, the eccentric error and the like of the encoder are effectively compensated.

The disclosure generally relates to a method, system and apparatus to improve a user's understanding of speech in real-time conversations by processing the audio through a neural network contained in a hearing device. The hearing device may be a headphone or hearing aid. In one embodiment, the disclosure relates to an apparatus to enhance incoming audio signal. The apparatus includes a controller to receive an incoming signal and provide a controller output signal; a neural network engine (NNE) circuitry in communication with the controller, the NNE circuitry activatable by the controller, the NNE circuitry configured to generate an NNE output signal from the controller output signal; and a digital signal processing (DSP) circuitry to receive one or more of controller output signal or the NNE circuitry output signal to thereby generate a processed signal; wherein the controller determines a processing path of the controller output signal through one of the DSP or the NNE circuitries as a function of one or more of predefined parameters, incoming signal characteristics and NNE circuitry feedback.

The disclosure generally relates to a method, system and apparatus to improve a user's understanding of speech in real-time conversations by processing the audio through a neural network contained in a hearing device. The hearing device may be a headphone or hearing aid. In one embodiment, the disclosure relates to an apparatus to enhance incoming audio signal. The apparatus includes a controller to receive an incoming signal and provide a controller output signal; a neural network engine (NNE) circuitry in communication with the controller, the NNE circuitry activatable by the controller, the NNE circuitry configured to generate an NNE output signal from the controller output signal; and a digital signal processing (DSP) circuitry to receive one or more of controller output signal or the NNE circuitry output signal to thereby generate a processed signal; wherein the controller determines a processing path of the controller output signal through one of the DSP or the NNE circuitries as a function of one or more of predefined parameters, incoming signal characteristics and NNE circuitry feedback.

The disclosure generally relates to a method, system and apparatus for processing audio through a neural network contained in a hearing device. In one embodiment, the disclosure relates to an apparatus to enhance incoming audio signal. The apparatus includes a controller to receive an incoming signal and provide a controller output signal; neural network engine (NNE) circuitry in communication with the controller, the NNE circuitry activatable by the controller, the NNE circuitry configured to generate an NNE output signal from the controller output signal; and digital signal processing (DSP) circuitry to receive one or more of controller output signal or the NNE circuitry output signal to thereby generate a processed signal; wherein the controller determines a processing path of the controller output signal through one of the DSP or the NNE circuitries as a function of one or more of predefined parameters, incoming signal characteristics and NNE circuitry feedback.

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.

A system and method for detecting anomalies in encrypted network traffic without decrypting encrypted payloads is disclosed. The invention comprises a hardware-accelerated pipeline including a packet acquisition unit configured to extract observable header attributes and timestamp values, a signal analysis unit implemented using digital signal processing circuitry to generate temporal and spectral characteristics of encrypted packet flows, and a morphometric extraction unit that produces encrypted-flow descriptors based on statistical dispersion, burst patterns, and ciphertext variability. A behavioral graph construction unit generates graph-structured representations of encrypted flows using incremental state-transition encoding. An artificial-intelligence inference unit employing a tensor-processing processor performs multi-stage anomaly analysis through temporal reconstruction, attention-based weighting of descriptors, and graph-structural deviation assessment to compute anomaly scores without decrypting any payload. A secure response control unit evaluates the scores against policy thresholds and performs mitigation actions while preserving confidentiality.

The present disclosure involves a medical device. The medical device includes an elongate cable assembly having a distal connector, a proximal connector, and a cable body coupling the proximal and distal connectors. The distal connector is configured for coupling with a diagnostic medical device. The proximal connector is configured for coupling with a medical measurement system. An electronic component is located inside the distal connector or the associated cable housing. The electronic component includes an analog-to-digital converter (ADC) and a microprocessor. The ADC is configured to receive medical data gathered by the diagnostic medical device and convert the medical data into digital signals. The microprocessor is coupled to an output of the ADC and configured to process the digital signals into a format that is readable by the medical measurement system.

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.

A system for material detection and identification includes an interface configured to access a material database associating each of a plurality of materials with a resonance frequency and one or more transmission parameters; an RF transmitter configured to, for each material of at least a subset of the plurality of materials in the material database transmit into an environment an RF signal the resonance frequency for the material using the one or more transmission parameters; an RF receiver configured to receive a response signal from the environment; and a processor configured to analyze the response signal for resonance characteristics that indicate a presence of the material and identify the material to a user if the presence of the material is indicated by the resonance characteristics.

The disclosure generally relates to a method, system and apparatus for processing audio through a neural network contained in a hearing device. In one embodiment, the disclosure relates to an apparatus to enhance incoming audio signal. The apparatus includes a controller to receive an incoming signal and provide a controller output signal; neural network engine (NNE) circuitry in communication with the controller, the NNE circuitry activatable by the controller, the NNE circuitry configured to generate an NNE output signal from the controller output signal; and digital signal processing (DSP) circuitry to receive one or more of controller output signal or the NNE circuitry output signal to thereby generate a processed signal; wherein the controller determines a processing path of the controller output signal through one of the DSP or the NNE circuitries as a function of one or more of predefined parameters, incoming signal characteristics and NNE circuitry feedback.

In order to solve the problem that quick response and distortionless recording of ultrahigh-frequency partial discharge signals are difficult to realize when accidental pulses are generated in the prior art, the invention relates to a dual-channel parallel processing architecture for ultrahigh-frequency partial discharge signals, which comprises a signal cloning circuit and a signal processing circuit, the ultrahigh-frequency partial discharge module is used for receiving ultrahigh-frequency partial discharge signals and synchronously outputting the ultrahigh-frequency partial discharge signals as first target signals and second target signals after lossless copying; the low-gain channel circuit is used for performing buffer transmission for maintaining an original bandwidth and a dynamic range on the first target signal to generate a first gain signal; the high-gain channel circuit is used for performing gain amplification on the second target signal to generate a second gain signal; the digital signal processing unit is used for performing synchronous clock control on the low-gain channel circuit and the high-gain channel circuit; and the FPGA chip is integrated in the digital signal processing unit and is used for performing fusion splicing processing on the first gain signal and the second gain signal based on a waveform fusion splicing algorithm and generating a fusion signal.

The invention requests to protect a multifunctional FFT (Fast Fourier Transform) hardware acceleration architecture capable of configuring software in real time, which comprises a first sample sequence rearrangement module, a first calculation unit, a second sample sequence rearrangement module, a second calculation unit, a third sample sequence rearrangement module, a butterfly operation circuit, a multiplexer and a dual-port RAM (Random Access Memory) which are sequentially connected end to end, in order to solve the technical problems that in an existing FFT hardware architecture, the operation length is fixed, the function is single, the fftshift function is lacked, and software dynamic configuration cannot be achieved, through software and hardware collaborative design, FFT hardware has operation flexibility, functional diversity and frequency spectrum processing capacity; the method can be widely applied to various digital signal processing scenes such as communication, images, radars and audios.

The present invention discloses a pulse generator and a pulse rod, which relate to the field of pulse circuit technology. The pulse generator includes a frequency control unit, a dual closed-loop pulse amplification unit, a digital signal processing unit and an adaptive power management unit. The frequency control unit includes an environmental sensor and a microcontroller. The microcontroller can calculate a preset pulse frequency based on information transmitted by the environmental sensor. The input end of the dual closed-loop pulse amplification unit can receive a pulse signal transmitted by the microcontroller, amplify it to a preset pulse frequency, and provide a pulse current to an electrostatic elimination needle. The digital signal processing unit can digitize the pulse signal to enhance the anti-interference ability of the pulse signal. The adaptive power management unit includes a dynamic voltage regulation module electrically connected to an external power supply, and the dynamic voltage regulation module can dynamically adjust the output voltage. The technical solution provided by the present invention can provide a pulse generator with high stability, a wide adjustment range and strong anti-interference ability.

The invention discloses an electric energy meter abnormal phenomenon automatic detection and identification method and system, and relates to electric digital data processing, and the method comprises the steps: carrying out the digital signal processing of initial data, the voltage effective value, the total harmonic distortion rate and the analysis of communication message abnormal codes, and generating a structured feature matrix; on the basis of the structured feature matrix, threshold comparison, composite condition judgment and continuous anomaly detection are executed through multi-level rule triggering, and an anomaly event report is generated; comparing the abnormal event report with a historical case, obtaining the similarity between the current abnormal event report and the historical case, and obtaining a similarity TOP-N case set; generating an anomaly diagnosis report by using weighted evidence fusion in combination with rule triggering intensity quantification and case matching credibility conversion; and updating the case library according to the abnormality diagnosis report, and counting and adjusting a rule threshold based on a sliding window. According to the method, the abnormal judgment threshold values of key parameters such as voltage and harmonic waves can be adjusted in real time through dynamic threshold value optimization, and the adaptability is improved.

The disclosure generally relates to a method, system and apparatus to improve a user's understanding of speech in real-time conversations by processing the audio through a neural network contained in a hearing device. The hearing device may be a headphone or hearing aid. In one embodiment, the disclosure relates to an apparatus to enhance incoming audio signal. The apparatus includes a controller to receive an incoming signal and provide a controller output signal; a neural network engine (NNE) circuitry in communication with the controller, the NNE circuitry activatable by the controller, the NNE circuitry configured to generate an NNE output signal from the controller output signal; and a digital signal processing (DSP) circuitry to receive one or more of controller output signal or the NNE circuitry output signal to thereby generate a processed signal; wherein the controller determines a processing path of the controller output signal through one of the DSP or the NNE circuitries as a function of one or more of predefined parameters, incoming signal characteristics and NNE circuitry feedback.

A radar system and control method thereof is disclosed. The radar system comprises a plurality of radar units, each comprising: one or more radio frequency (RF) channels configured to receive a reflected signal and then generate an analog input signal according to the reflected signal; and a processing module connected with all the RF channels and configured to sample the analog input signal to obtain a digital signal and perform the first digital signal processing on the digital signal to obtain intermediate data, wherein when the plurality of radar units work jointly, a designated radar unit performs the second digital signal processing on the plurality of intermediate data provided by the plurality of radar units, thereby obtaining result data of the radar system.

The invention relates to a detection-interference integrated unmanned aerial vehicle electromagnetic interference device, which belongs to the technical field of electronic countermeasure, and comprises an integrated design of an antenna module, a radio frequency front end module, a digital signal processing module and a main control module, the radio frequency front-end module comprises a detection receiving channel and an interference transmitting channel to process signal conversion, the digital signal processing module executes signal detection and analysis, intelligent interference decision and anti-self-interference signal processing based on a self-adaptive filtering algorithm, and the main control module is responsible for overall control and power supply. According to the invention, the technical problems of large response delay, complex system cooperation and incapability of continuous reconnaissance in the interference process of traditional anti-unmanned aerial vehicle equipment are solved. Real-time signal cancellation is achieved through the normalized least mean square algorithm, the reconnaissance capability is kept while interference is applied, the method has the advantages of being rapid in response, accurate in interference, low in power consumption and the like, and the countering efficiency is remarkably improved.

The invention relates to an audio SOC chip, an audio subsystem interface module thereof, and electronic equipment, and the audio subsystem interface module comprises a plurality of downlink path units which are used for carrying out the downlink processing of at least one path of digital audio data obtained from an application processor chip and / or a digital signal processing chip through a peripheral bus interface module, the audio data is output through the audio data transmission interface module; the plurality of uplink path units are used for performing uplink processing on at least one path of digital audio data input through the audio data transmission interface module and outputting the processed data to the application processor chip and / or the digital signal processing chip through the peripheral bus interface module; and the plurality of loopback path units are used for converging the at least one path of digital audio data input through the audio data transmission interface module and the digital audio data obtained by performing downlink processing through any downlink path unit in the plurality of downlink path units, and outputting the digital audio data through the audio data transmission interface module, so that the power consumption of a chip is reduced.

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 discloses a data sampling timing error evaluation method based on an evaluation model, relates to the technical field of timing error evaluation, and solves the problem that timing error evaluation is not accurate enough. Wherein the sampling module realizes undistorted receiving and clock signal frequency output of a clock signal through a CMOS clock input circuit, a third-order digital phase-locked loop circuit and a DSP digital signal processing chip, and a Fourier phase difference algorithm removes harmonic interference to solve a phase difference between the clock signal and an ideal clock signal; the clock signal feature recognition unit adopts a convolutional neural network to carry out iterative training on a training set of the timing error evaluation module to obtain an error evaluation module, the error evaluation module and the time migration algorithm module realize accurate calculation of a timing error value, and the timing error value calculation accuracy is greatly improved.

The invention provides a portable micro-beam X-ray fluorescence rapid detection method and system based on digital signal processing. The method comprises the following steps: firstly, constructing a micro-beam excitation parameter adaptive to a field environment, secondly, driving a micro-beam X-ray to irradiate a to-be-detected sample by using the micro-beam excitation parameter, synchronously obtaining an original fluorescence signal disturbed by the environment, then taking a dynamic adjustment amount as a calibration reference, and fusing the calibration reference and the original fluorescence signal to generate a combined data stream; then carrying out cooperative processing on the combined data stream through a digital signal processing technology, separating an environment disturbance component from a target fluorescence component, and finally generating a sample surface element distribution map based on the target fluorescence component. According to the technical scheme provided by the invention, the problems of morphology misalignment and dynamic noise interference caused by passive noise reduction design are solved, and the element detection reliability of complex morphology samples (such as soil blocks and cultural relic sections) is also improved.

The invention discloses a motor position non-contact sensing system and a signal processing control method. The system comprises a three-dimensional magnet assembly, a magnetic signal acquisition module, a digital signal processing unit, a voltage conversion unit and a cooperative control module. The three-dimensional magnet assembly and the acquisition module are coaxially arranged, magnetic points are in spatial gradient layout, the acquisition module comprises a three-dimensional magnetic field detection unit, a real-time phase calibration unit is arranged in the processing unit, and the cooperative control module adjusts sampling parameters based on the rotating speed. The method comprises the steps of initialization, magnetic signal acquisition, operation processing and voltage conversion output. According to the scheme, the problem of magnetic field signal distortion of an existing system is solved, stable and accurate sensing of the motor position is achieved, and the multi-scene motor control requirement is met.

The invention discloses a channel-configurable liquid particle pulse amplitude detection method and a related device, and relates to the field of digital signal processing, and the method comprises the following steps: eliminating a background noise reference value of a digital pulse signal, presetting a dynamic threshold value, filtering a noise signal of which the amplitude value is lower than the dynamic threshold value in the digital pulse signal, and obtaining a channel-configurable liquid particle pulse amplitude detection result; obtaining a baseline correction digital pulse signal; when the amplitude of the baseline correction digital pulse signal exceeds a dynamic threshold for the first time, pulse counting is started, peak detection is carried out on the baseline correction digital pulse signal, and a pulse peak value and a peak value position corresponding to a pulse event are obtained; a channel threshold parameter is preset through an upper computer, a plurality of independent channels are divided, pulse events are loaded into the corresponding independent channels to be accumulated, a channel count value and corresponding particle size distribution data are obtained, and liquid particle pulse amplitude detection is completed. The problem that the number of channels and threshold parameters cannot be dynamically adjusted when a current hardware circuit is used for liquid particle pulse amplitude detection is solved.

The invention discloses a multi-voice common exchange type anti-interference Bluetooth earphone translation system, and relates to the technical field of voice processing. Comprising an audio dynamic state acquisition module, an audio feature preprocessing and data construction module, a dynamic range abnormal feature extraction and quantitative analysis module, an audio system state intelligent evaluation module and a sound source weight adaptive regulation and control module, through a high-precision microphone array, a digital signal processing chip and an audio processing module which are integrated in the earphone, audio dynamic range state characteristic data generated in the operation process of each sound source channel are monitored and captured in real time. According to the method, the audio dynamic range imbalance state is recognized in real time, the sound source weight is adjusted in a self-adaptive mode, interference channel local suppression and main voice channel stable maintenance are achieved, and translation task continuity and recognition accuracy are effectively guaranteed.

A zonal audio system for a vehicle may include a microcontroller unit (MCU), a non-volatile storage communicatively coupled with the MCU that stores audio files for different types of audio functions, and a digital signal processing (DSP) module communicatively coupled with the MCU. The DSP module may process multiple types of audio functions simultaneously through a layered architecture that consolidates and prioritizes different audio streams including emergency calls, acoustic vehicle alerts, chimes, and infotainment.

The invention provides a polarization diversity and frequency diversity combined low-altitude communication anti-interference system and a channel allocation method, and relates to the technical field of low-altitude communication, and the system is composed of a multi-polarization antenna array, a broadband radio frequency front end, a digital signal processing unit, a self-adaptive controller and an environment sensing module. The core of the invention is a polarization-frequency joint channel allocation method, and global optimal allocation of channel resources is realized by constructing a polarization-frequency two-dimensional channel quality matrix. The system adopts a multi-layer adaptive architecture and comprises a quick response layer and a deep optimization layer, so that the real-time performance and the optimization degree are balanced. In addition, the invention also provides an anti-intelligent interference diversity parameter dynamic adjustment mechanism and an interference identification method based on polarization characteristics, so that the communication reliability and the anti-interference capability of the system in a complex electromagnetic environment are remarkably improved. And communication with an extremely low bit error rate can still be kept in a strong interference environment.

The invention relates to a high-precision hardware elastic sheet punch forming method in the field of high-end equipment manufacturing industry, which comprises the following steps: acquiring positioning data in a punching process, acquiring real-time displacement data of a driving system and a die through a high-precision sensor, and denoising the acquired data by adopting a digital signal processing technology to obtain a first positioning data set; a finite element analysis technology is adopted to simulate the contact state of the mold and a material, the influence weight of mold design parameters on positioning precision is obtained, and an optimized mold geometric parameter data set is determined; according to the optimized mold geometric parameter data set, a digital model of the multi-station mold is generated, inter-process errors of the mold under multi-station continuity are verified through a simulation technology, and a process error data set is obtained; the method comprises the following steps: acquiring stress distribution data of a material in a stamping process, acquiring a real-time stress value through a strain sensor, and performing staged control on the stamping process by adopting a progressive stress release algorithm to obtain a first stress distribution data set;