47 results about "Iir filtering" patented technology

The invention discloses a coupling pulse distortion compensation method for superconducting quantum computing, and relates to the field of signal processing, and the method comprises the steps: S1, inputting an initial square wave signal into a first IIR filter, and reducing the frequency attenuation of superconducting quantum bits; s2, outputting the initial square wave signal to a second IIR filter to improve the high-frequency component of the initial square wave signal; s3, outputting the initial square wave signal to a third IIR filter to eliminate pulse back edge reflection of the initial square wave signal; and S4, outputting the initial square wave signal to a first FIR filter to carry out ripple correction so as to complete pre-distortion correction. By setting the staged pre-distortion compensation of the three-section IIR filter and the one-section FIR filter, the pre-distortion compensation of the square wave signal is more accurate and stable, the square wave edge response is rapid and the amplitude is correct, and the time delay and the amplitude error caused by the link are reduced; the quantum bit Z-axis phase control method has the advantages that quantum bit Z-axis phase control is more accurate, and the fidelity of a square wave signal gate is prevented from being reduced.

The application relates to a processor, component, device and method for filter processing of an IIR filter. The processor comprises a first configuration register, a second configuration register, a general register and a matrix multiplication accumulation unit. The general register sequentially reads and stores each row of coefficients of a coefficient matrix of each order of the IIR filter. The matrix multiplication accumulation unit obtains the same input element corresponding to the stored current row of coefficients when the first configuration register is configured in a broadcast mode, copies the stored current row of coefficients when the first configuration register is configured in a copy mode, multiplies the current row of coefficients with the corresponding input element in parallel to obtain corresponding products, and sequentially accumulates the product results of each row of coefficients to obtain a final output value, so that at least four output variables of sequential sampling moments are obtained each time. In this way, the time consumption of IIR filter operation can be significantly shortened, and sufficient acceleration effect is provided.

The present application belongs to the technical field of medical ultrasonic signal processing, and specifically relates to a spermatic vein ultrasonic imaging enhancement system and method based on double-wall filtering and adaptive Doppler signal processing. The system comprises a dynamic reconfigurable analog front end, a dual-channel heterogeneous data processing unit and a closed-loop feedback controller. The method extracts fluid dynamics characteristics in real time through an auxiliary path, intelligently identifies resting state and action mutation state. At the action mutation moment, the physical layer automatically switches the analog filtering parameters to prevent signal saturation, and the digital layer adaptively increases the FIR filter weight to eliminate the ringing artifact; in the resting state, the high-gain analog configuration is maintained and the IIR filter is dominated to capture weak signals. With the delay buffer mechanism, accurate timing alignment of control instructions and imaging data is realized. The present application effectively solves the contradiction between low flow rate detection sensitivity and strong tissue motion interference suppression, eliminates the flicker artifact generated by breath holding action, and significantly improves the diagnosis accuracy of weak reflux.

The invention discloses a vehicle-mounted audio equalization method and system based on residual vector quantization, and relates to the technical field of vehicle-mounted electronic equipment.The method comprises the steps that a white noise audio signal is played through a vehicle-mounted loudspeaker, a microphone is used for collecting the audio signal lost through in-vehicle propagation, and time-frequency characteristics are obtained through digital preprocessing; performing discretization processing on the target IIR filter coefficient through a residual vector quantization encoder to generate a plurality of layers of token sequences; inputting the time-frequency feature matrix into a convolutional neural network for feature extraction to obtain a low-dimensional feature vector, and through a multi-layer perceptron prediction head, respectively obtaining classification probability distribution of each layer of token and predicting a token sequence; and inputting the predicted token sequence into a residual vector quantization decoder, reconstructing the token sequence into an IIR filter coefficient, loading the IIR filter coefficient into a graphic equalizer, and performing frequency response compensation on an input audio signal. According to the method, the audio frequency response curve can be quickly and adaptively corrected, and the in-vehicle tone quality and the user hearing comfort are remarkably improved.

A system for converting from a finite impulse response (FIR) filter to an infinite impulse response (IIR) filter, includes: a processor and a memory, including instructions stored thereon, which when executed by the processor cause the system to: access a FIR filter representative of a physical space; access a personalized Head Related Impulse Responses (HRIRs); generate a fused FIR filter based on fusing the FIR filter and the personalized HRIR; determine an order of an IIR filter based on the fused FIR filter; generate the IIR filter based on the determined order; and optimize the generated IIR filter by performing gradient descent minimization to minimize a cost function of the generated IIR filter.

An audio sampler is provided, comprising a sample library having stored therein a plurality of main audio samples and a plurality of infinite impulse response (IIR) coefficients divided into subset, each subset corresponding to one of the main audio samples; a sample playback engine comprising an input receiving the main audio samples and outputting a playback signal; and, an IIR filter receiving the corresponding subset of IIR coefficients and applying the IIR filter to the main audio samples or to the playback signal.

The application discloses an audio loudness improvement and dynamic range optimization method and related device, and the method comprises the following steps: obtaining an audio signal to be optimized, performing frequency response correction and dynamic noise suppression filtering processing on the audio signal to be optimized based on a multi-stage IIR filter, and obtaining a filtered audio signal; designing an adaptive dynamic compressor based on an LUFS loudness evaluation model, performing dynamic compression processing on the filtered audio signal by using the adaptive dynamic compressor, and obtaining a compressed audio signal; performing filtering optimization processing on the compressed audio signal based on a multi-stage low-pass filter, and outputting a filtered and optimized audio signal. In the embodiment of the application, the loudness is improved, the dynamic details of the audio signal are retained, the high-frequency noise amplification and the excessive compression of the low-frequency signal are avoided, the automatic dynamic compression of the audio signal is realized, and the problems such as "breathing effect" and "peak leakage" are eliminated.

The present disclosure relates to a kind of energy storage converter multi-mode control method and device.Wherein, the method includes: for energy storage converter in grid-connected mode, by creating second-order infinite impulse response IIR filter, establish low voltage ride through control strategy based on compliance repeat control;For energy storage converter in off-grid mode, three-phase four-bridge topology is used to provide zero sequence path, establish preset proportional resonant control strategy;For energy storage converter in grid-constructing mode, establish parallel pre-synchronization control strategy;Low voltage ride through control strategy based on compliance repeat control, preset proportional resonant control strategy, parallel pre-synchronization control strategy are used to realize multi-mode control to the energy storage converter.The present disclosure can set any resonant frequency and filter radius to suppress the resonant peak, ensure system stability, compared with traditional pre-synchronization method only using its frequency modulation characteristics, without additional control link can complete pre-synchronization, greatly save control resources.

The application discloses a polar region adaptive underwater acoustic positioning correction method based on a multi-dimensional environment coupling model, belongs to the technical field of underwater acoustic positioning, and is used for polar region underwater acoustic positioning and comprises the following steps: deploying a heterogeneous sensor network, constructing an environment perception layer, and eliminating sensor drift errors; a three-dimensional sound velocity model is established, a vortex dynamics equation, a Kriging interpolation algorithm and an adaptive Runge-Kutta method are combined to track a sound ray track; a time-frequency dual-domain joint detection algorithm is designed, a lattice IIR filter bank is combined to realize multipath suppression; a 9-dimensional robust UKF algorithm framework is constructed, and multi-source data fusion is realized in combination with federal filtering; and a geomagnetic matching and strain energy density early warning mechanism is integrated. Compared with the prior art, the application improves the positioning accuracy and stability, can more accurately predict a sound propagation path, improves positioning precision, realizes dynamic real-time calculation, can quickly and accurately process changing environmental data, and is suitable for complex polar region environments.

This invention provides a method for processing echo signals from a fish finder, the fish finder itself, and a storage medium, comprising: S1, receiving an echo signal; S2, performing envelope processing on the echo signal to obtain an envelope signal; S3, calculating the variance of data within a selected sliding window until a signal segment with a variance value less than a predetermined variance threshold is found; S4, determining the noise floor value of the envelope signal based on the average amplitude of the found signal segment; S5, determining the starting position of a weak signal data segment based on the location of signal points with amplitudes less than the noise floor value; S6, performing partial filtering on the envelope signal to obtain the image signal to be displayed, and performing IIR filtering or FIR filtering on the weak signal data segment; signal segments other than the weak signal data segment retain their original values. Using the above technical solution, weak signals in the echo signal can be identified, thereby improving the sensitivity of the fish finder.

A method and system for suppressing channel response unevenness based on OFDM signals, including: channel response unevenness estimation, compensation, the channel response unevenness estimation including: converting a signal into an amplitude-phase method representation, channel estimation, IIR filtering, updating a signal estimate; the method and system for suppressing channel response unevenness based on OFDM signals, through filtering operation, reduces the risk of performance sharp decline when an individual carrier is suddenly affected by fading or noise, reduces the burr phenomenon from the flatness result, makes the flatness result closer to the power amplifier frequency domain property of the DUT, and suppresses the performance calculation deterioration caused by carrier mutation, which helps to improve the decoding ability to a certain extent.

This invention discloses an adaptive filtering method and system for ultrasonic water meter flow measurement. The adaptive filtering method for ultrasonic water meter flow measurement includes: S1. Obtaining the original measurement value of the time difference between upstream and downstream flight of the ultrasonic water meter; S2. Calculating the absolute difference between the current original measurement value and the previous original measurement value, comparing the absolute difference with a preset anomaly detection threshold, and proceeding to step S3 when the absolute difference is less than the preset anomaly detection threshold; proceeding to step S4 when the absolute difference is greater than or equal to the anomaly detection threshold; S3. Determining that the current original measurement value is a normal fluctuation, maintaining the current filtering coefficient or gradually increasing the filtering coefficient, and performing filtering output according to a first-order IIR filtering algorithm; S4. Determining that the current original measurement value is an abnormal sudden change, performing first-order IIR filtering output on the current original measurement value using a preset minimum filtering coefficient.

The application belongs to the technical field of controllable phase noise generation, and particularly relates to a self-defined phase noise adjusting method and system based on an IIR filter. The method comprises the following steps: S1, power collaborative allocation; S2, Gaussian white noise generation; S3, noise frequency response modeling based on the IIR filter; S4, noise shaping; S5, dynamic resampling adaptation; and S6, converting the digital domain noise vector signal into an analog radio frequency signal output through a vector modulator and a DAC. The application innovatively combines the noise frequency response precise modeling based on the IIR filter, the dynamic resampling adaptation mechanism, the noise shaping and the power collaborative allocation algorithm, realizes self-defined phase noise adding of the original signal, simulates the phase noise of an actual oscillation circuit, and has low power consumption and high data throughput capacity, thereby meeting the strict timing requirements of high-speed real-time signal processing, and being particularly suitable for real-time signal processing scenes.

The application discloses a pigging indication algorithm using a pigging indication device based on sound waves, comprising a data acquisition and processing unit, an explosion-proof box and a pigging indication algorithm module; the data acquisition and processing unit comprises an acceleration sensor group, a safety grid group, an A / D conversion unit, a control unit, a WiFi transmission unit, a data storage unit and a battery group. The application designs an explosion-proof structure for the potential explosion of oil and gas pipelines; on the other hand, the sound wave signals in the pipeline are preprocessed and the pigging time is calculated. Among them, for various working condition interference sound wave signals in the pipeline, IIR filtering, data scaling and feature extraction are adopted to suppress the interference signals; for the processed pipeline sound wave signals, the time corresponding to the maximum value of two features is extracted, and weighted summation is carried out to realize pigging alarm and pigging time calculation.

The invention provides a second-order IIR filter. The second-order IIR filter comprises a second-order IIR filtering unit, a control unit and an output unit. The control unit determines whether the filter coefficient needs to be updated according to the frequency information of the input signal, calculates the latest filter coefficient based on the frequency information of the input signal when the filter coefficient needs to be updated, and provides the latest filter coefficient for the second-order IIR filtering unit for updating; and the second-order IIR filtering unit carries out filtering processing on the input signal based on the latest filter coefficient immediately after the filter coefficient is updated, and obtains a filtering output signal. And the output unit continuously takes the last filtering output signal output by the second-order IIR filtering unit before the filter coefficient is updated as the output signal of the second-order IIR filter at N output moments after the filter coefficient is updated. In this way, noise is prevented from occurring in the output signal.

The application discloses a dense false target generation method based on segmentation reconstruction, which is applied to an FPGA provided with an IIR filter, a feedback branch of the IIR filter is provided with a register, and a forward branch of the IIR filter is connected with a RAM, and the method comprises the following steps: saving signals collected by a DRFM into the RAM; according to a minimum segmentation length, signal segments at each moment are sequentially obtained from the DRFM, and after register delay and accumulation, corresponding accumulated values at each moment are obtained; after the obtained signal segments reach a given false target number M, signal segments before M moments at each moment are sequentially obtained from a starting position of the external RAM according to the minimum segmentation length, and the corresponding accumulated values at each moment are subtracted from corresponding signal segment data obtained from the external RAM, and the corresponding accumulated values at each moment are taken as data of signal segments at each moment of the dense false target signal. The application overcomes the problem that the segmentation reconstruction method consumes too much storage resource when the total delay length is large.

The invention relates to a sound field balancing method and device for sound waves in a vehicle, electronic equipment, a medium and a product. Comprising the steps that preset time domain noise is played through a plurality of loudspeakers of a vehicle with a sound field to be balanced, sound pressure signals are collected through first to fourth collection pieces, and corresponding to fourth frequency response amplitudes at a plurality of preset center frequencies are extracted; calculating a difference value between the maximum frequency response amplitude and the minimum frequency response amplitude at each center frequency, and identifying a target center frequency of which the difference value is greater than a preset threshold value; and determining target parameters of each loudspeaker IIR filter according to the frequency response amplitude at the target center frequency, adjusting the corresponding parameters based on the preset sampling frequency, and iteratively calculating the standard deviation of the frequency response amplitude until the difference values at all the target center frequencies are smaller than or equal to a preset threshold value. Therefore, the problems that sound fields of four positions including the front row and the rear row in the vehicle are not uniform, and particularly when the acoustic performance of the whole vehicle is poor, sound wave balance is difficult to achieve are solved, and the driving experience of a user is greatly improved.

A digital microphone incorporates an analog-to-digital converter (ADC) and a digital filter system coupled to the ADC, with the digital filter system switching between a standard IIR filter architecture and a polyphase IIR filter architecture.

The invention provides a second-order IIR (Infinite Impulse Response) filter and an optimization method thereof. The optimization method comprises the following steps: for a target cut-off frequency fc, setting a target cut-off frequency interval as [fcmin, fcmax], and setting a frequency interval delta f; generating a list formed by a plurality of candidate target cut-off frequencies in the target cut-off frequency interval by taking a frequency interval delta f as a step length, wherein the ith candidate target cut-off frequency is fc (i) = fcmin + i * delta f; calculating a filter system corresponding to each candidate target cut-off frequency, quantizing a filter coefficient corresponding to each candidate target cut-off frequency to obtain a quantized filter coefficient, and calculating a quantization error of a second-order IIR filter based on the quantized filter coefficient; and selecting the candidate target cut-off frequency with the minimum quantization error as the actual target cut-off frequency, and using the quantized filter coefficient corresponding to the actual target cut-off frequency as the filter coefficient of the second-order IIR filter. In this way, the quantization error of the whole system is reduced by optimizing the quantization process of the filter coefficient.

The invention discloses a reconfigurable digital filter based on an FPGA (Field Programmable Gate Array), which comprises an FPGA chip, and the FPGA chip comprises an ROM (Read Only Memory) module, an RAM (Random Access Memory) module, a multiplier, an accumulator, a pulse counter, a filtering control module and a reconfiguration interface, the inertial device comprises at least one gyroscope and an accelerometer, and the gyroscope and the accelerometer both output pulse signals to the FPGA chip; the pulse counter counts the pulse signals, converts count values of the N channels into floating-point numbers after the counting period is ended, and stores the floating-point numbers into the RAM module; the filtering control module controls the multiplier, the accumulator, the ROM module and the RAM module to work cooperatively, multiplexes the same group of computing resources to process signals of N channels in series, and completes filtering calculation according to an IIR filter algorithm; the reconfiguration interface can adjust the order and coefficient of the IIR filter online. The filter replaces DSP filtering, the FPGA resource utilization rate is increased, the chip cost is reduced, the execution efficiency of an inertial navigation system is improved, and online reconstruction can be carried out to adapt to different filtering requirements.

An IIR filtering component, an audio device and a coefficient switching method. In the case where it is necessary to adjust the filtering coefficients used for filtering processing on an input signal, the weights of the filtering coefficients of two sets of filtering resources are adjusted, that is, the output signal component proportion of the source filtering coefficient and the output signal component proportion of the target filtering coefficient in the filtering output signal are adjusted, so that the filtering output signal is gradually adjusted from the output corresponding to the source filtering coefficient to the output corresponding to the target filtering coefficient.

The application provides an automatic and accurate correction method and system of an acoustic equalizer, and relates to the technical field of acoustic control. The method comprises the following steps: obtaining a standard audio signal; obtaining a secondary test signal; obtaining a fitting target signal; obtaining a correction signal based on the standard audio signal, the secondary test signal and the fitting target signal; obtaining a center frequency sequence, a compensation center frequency sequence, a gain coefficient sequence, a compensation gain coefficient sequence and a bandwidth information sequence of the equalizer; calculating filter coefficients and a frequency response curve of a filter; obtaining an actual optimal gain sequence, calculating correction filter coefficients; and inputting the correction filter coefficients into a second-order IIR filter to realize accurate correction of a sound field. The technical problems of the prior art, such as the difficulty in avoiding the interference of human factors, the ease in producing subjective deviation and the low efficiency, are solved. The technical effect of the acoustic equalizer, such as high efficiency, high precision, no interference of human factors and support for personalized adjustment, is achieved.

According to an aspect, an apparatus may comprise a digital pre-distorter, DPD, circuitry including at least one first-order infinite impulse response, IIR, filter and at least one second-order IIR filter. The apparatus may further comprise a power amplifier, at least one processor and at least one memory comprising instructions which, when executed by the at least one processor, cause the apparatus at least to compensate for non-linearities in the power amplifier by using the at least one first-order IIR filter and the at least one second-order IIR filter.

The invention relates to the technical field of filter circuits, and discloses a filter circuit for an ADC, a true RMS calculation circuit and the ADC, and the filter circuit comprises an analog modulator, an IIR filter and a CIC filter. The analog modulator is used for modulating an input analog signal and outputting a sparse and dense wave signal; the IIR filter is used for performing medium-high frequency filtering on the sparse and dense wave signal when the OSR of the analog modulator is lower than a threshold value, and outputting a primary filtering signal; the CIC filter is electrically connected with the IIR filter and is used for filtering the primary filtering signal and outputting a secondary filtering signal; when the filter circuit is used, the IIR filter and the CIC filter are arranged, and the IIR filter is used for performing medium-high frequency filtering on sparse and dense wave signals when the OSR of the analog modulator is lower than a threshold value, so that medium-high frequency noise can be filtered preliminarily, signal attenuation is more ideal, attenuation pressure of the CIC filter is reduced, signal mixing can be reduced, and the filter circuit has the advantages of being simple in structure, low in cost and the like. And the stop-band suppression problem existing when the OSR is relatively low is avoided.

The application discloses an intelligent control system and method of an out-of-vehicle handle capable of preventing false triggering, and relates to the technical field of automobile electronics. In the application, an electrode array is arranged in a detection area of an out-of-vehicle handle, and self-capacitance and mutual-capacitance detection are alternately performed to collect capacitance signals. The capacitance signals are subjected to median filtering and IIR filtering processing, and then normalized by using a dynamic reference signal to obtain normalized self-capacitance and mutual-capacitance signals. The amplitude, change rate trend and fluctuation rate of the signals are comprehensively analyzed, and compared with preset threshold values, so as to accurately distinguish between human body touch and external interference, and perform corresponding operations. A clustering analysis mechanism is introduced to learn historical judgment results, automatically optimize the distinguishing threshold value, and form a closed-loop optimization. The application effectively overcomes the defects of false judgment of conductive media such as water in the traditional scheme, and significantly improves the accuracy and environmental adaptability of touch control recognition.

The invention discloses a deep learning-based operation power tool doctor operation intention identification method, which is characterized in that balance between a filtering effect and a response speed is improved based on an SSNR adaptive filtering switching mechanism, different noise spectrums are automatically adapted through online optimization of IIR filter coefficients, and manual readjustment is not needed; meanwhile, by introducing a Top-k multi-scale period extraction strategy, the sensing ability of the model to weak period signals is enhanced. The recognition method solves the problems that in the prior art, doctor operating force and tool counter-acting force are seriously coupled, high-strength vibration noise interference is remarkable when a power tool runs, and the real operation intention of a doctor cannot be accurately recognized.

The invention discloses a micro-vibration monitoring method based on foreground detection and adaptive Euler video amplification, and belongs to the technical field of equipment micro-vibration monitoring, and the method comprises the steps: firstly collecting an electromechanical equipment operation video, extracting a foreground through a Gaussian mixture model to obtain a mask, and segmenting the mask into sub-regions through an SLIC algorithm; 4 vibration indexes of each sub-region are calculated and quantized intensity is fused, so that the number of layers of the pyramid and an amplification coefficient are matched; and building a Gaussian pyramid, a Laplacian pyramid and double IIR filtering to extract a target vibration signal, and reconstructing an image after amplifying and superposing an original signal according to parameters to obtain a vibration enhanced video. According to the micro-vibration monitoring method based on foreground detection and adaptive Euler video amplification, precise monitoring of micro-vibration of equipment is realized, adaptive matching of amplification parameters is realized, loss of details and artifacts are avoided, the identification degree of vibration signals is improved, meanwhile, the method adapts to a complex industrial environment, a clear vibration enhanced video is output, and the method is suitable for popularization and application. Faults of the assisting equipment can be found early, and operation and maintenance efficiency and safety are guaranteed.

The invention discloses an FPGA-based time division multiplexing IIR filter element second-order section, which comprises a dual-port BRAM, a DSP module and a control module which are arranged in the FPGA, and is characterized in that the dual-port BRAM is used for storing IIR filter parameters corresponding to the current element second-order section and caching input data and output data, and allocating a register set for the element second-order section; the dual-port BRAM is used for caching input data, output data, a zero point coefficient and a pole coefficient, the DSP module carries out element second-order section calculation through an 11-beat time sequence calculation process, and the control module controls the dual-port BRAM and the DSP module to complete element second-order section calculation. According to the method, the hardware implementation mode of the element second-order section is improved, area resources in the FPGA are saved based on the thought of time division multiplexing, the resource utilization rate is increased, and the control complexity is reduced.

This invention relates to an adaptive time-frequency joint filtering method for electrostatic data from solid rocket motor exhaust. The method includes the following steps: constructing a main channel filtering algorithm to suppress the fundamental combustion oscillation, employing an adaptive IIR notch filter, with the core frequency detected by a real-time FFT algorithm and the bandwidth dynamically adjusted according to the combustion state; constructing an auxiliary channel to suppress harmonics, consisting of multiple parallel notch filters, each corresponding to a harmonic frequency, with the harmonic order adaptively determined based on the fundamental attenuation; addressing the phase distortion problem of traditional IIR filters by using forward-backward filtering to offset phase shift; particle pulse detection is the core of solid rocket motor exhaust electrostatic signal processing, aiming to accurately identify transient pulses (1-5 μs) generated by Al2O3 particle collisions; this invention uses a wavelet domain adaptive algorithm based on multi-scale noise separation, a time-varying threshold mechanism, and the SURE optimization criterion to suppress combustion oscillations and noise in the exhaust electrostatic signal while preserving particle pulse information.

The invention belongs to the technical field of automobile audio control, and particularly relates to an audio system low-frequency response dynamic compensation method and system in a vehicle driving state. The dynamic compensation method comprises the following steps: acquiring a vehicle speed signal when the vehicle speed is gt; when the speed is greater than the set speed, compensation is started; the method comprises the following steps: collecting sound signals in a vehicle, and extracting noise energy values of at least four critical frequency bands in a 20-150Hz frequency band through framing, windowing and FFT (Fast Fourier Transform); and when the frequency point energy of the environmental noise is higher than a threshold value, calculating the compensation gain of each frequency band, dynamically updating the gain parameter of the IIR filter bank, and performing frequency-band equalization processing on the input audio. According to the technical scheme, the music low-frequency component masked by the driving low-frequency noise is compensated in a targeted manner instead of simply and integrally increasing the low frequency, so that the music can be heard more clearly and more evenly during driving, the low-frequency details are richer, and the overall tone quality is better.