77 results about "Iir filtering" patented technology

The invention 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, is used for polar region underwater acoustic positioning, and comprises the steps of deploying a heterogeneous sensor network, constructing an environment sensing layer and eliminating a sensor drift error. Establishing a three-dimensional sound velocity model, fusing a vortex kinetic equation, a Kriging interpolation algorithm and an adaptive Runge-Kutta method to track a sound ray trajectory; designing a time-frequency double-domain joint detection algorithm, and combining a lattice IIR filter bank to realize multipath suppression; a 9-dimensional robust UKF algorithm framework is constructed, and multi-source data fusion is realized in combination with federated filtering; and terrestrial magnetism matching and a strain energy density early warning mechanism are integrated. Compared with the prior art, the method improves the positioning accuracy and stability, can more accurately predict the sound propagation path, improves the positioning precision, achieves the dynamic real-time calculation, can quickly and accurately process the continuously changing environment data, and is suitable for the complex polar region environment.

The invention discloses an IIR (Infinite Impulse Response) filter device and a filtering method thereof, and the IIR filter device comprises a head data module which stores current input data and two pieces of input historical data; the filtering module is used for storing output data of each stage through a first position, corresponding to each stage of filtering, of the intermediate data module, exchanging addresses of the first position and a second position, reading current input data and two pieces of input historical data from a specified position through the calculation module, and outputting the data; two pieces of output historical data are read from the intermediate data module and stored in the tail data module, data are output after filtering, and lower-level filtering is triggered; the tail data module is used for storing the output data of the level of filtering as the input data of the lower level of filtering when any level of filtering is completed; wherein the designated position of the first-stage filtering is a head data module, and the designated positions of the other-stage filtering are tail data modules. According to the method, the data reading and writing time is shortened, the calculation pipeline is prevented from being interrupted, the required data storage is reduced by half, and the required clock frequency is reduced.

The invention discloses a semi-active suspension control strategy based on IIR (Infinite Impulse Response) filter frequency division, which is characterized in that an acceleration sensor is used as input, a low-pass IIR filter and a high-pass IIR filter are respectively used for processing the acceleration of a cab, and appropriate damping is output according to a result so as to suppress the resonance phenomenon of a vehicle body during low-frequency vibration; and the vibration isolation coefficient during high-frequency vibration is improved. According to the method, the number of required sensors is small, the complexity is low, the computing power requirement for hardware is small, the cost of the semi-active suspension can be effectively reduced, and the method has a certain engineering practical application value and is high in generalizability. Test results show that compared with passive suspension, the control strategy can effectively reduce vibration of a vehicle cab in the x direction and the z direction, so that the comfort index of the long-distance logistics vehicle is improved, x-direction optimization is 21.3% at most, and z-direction optimization is 11.7% at most.

The invention provides an automatic accurate correction method and system for a sound equalizer, and relates to the technical field of sound regulation and control, and the method comprises the 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; acquiring 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 a filter coefficient and a frequency response curve of the filter; obtaining an actual optimal gain sequence, and calculating a correction filter coefficient; and inputting the correction filter coefficient to a second-order IIR filter to realize accurate correction of the sound field. The technical problems that in the prior art, interference of human factors is difficult to avoid, subjective deviation is likely to be generated, and efficiency is low are solved. The technical effects of high efficiency, high precision, no interference of human factors and capability of supporting personalized adjustment of the sound equalizer are achieved.

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 embodiments of the present application provide an active noise reduction method, device, active noise reduction headphones and storage medium, which relate to the field of audio noise reduction. A processor applied to an active noise reduction headphone, the active noise reduction headphone includes a feedforward active noise reduction filter, the feedforward active noise reduction filter includes an IIR filter and a FIR filter; the active noise reduction method includes: obtaining a current first transfer function value; adjusting the IIR filter to a target configuration corresponding to the current first transfer function value based on the current first transfer function value; estimating the current overall frequency response required for the feedforward active noise reduction filter under the current first transfer function value; determining the target frequency response of the FIR filter based on the target configuration and the current overall frequency response; adjusting the FIR filter based on the target frequency response to use the adjusted FIR filter for noise reduction. This method helps to reduce the chip area required for the feedforward active noise reduction filter.

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 provides an intelligent filtering control method for a hydrogen storage tank of a wind-solar coupled electrolytic hydrogen production system, which comprises the following steps: designing a filter, dynamically adjusting the cut-off frequency of the filter according to the time constant of the hydrogen storage tank, and finding the optimal filter order through an optimization mechanism; the current working condition is judged according to the hydrogen charging and discharging state of the hydrogen storage tank, and a proper type is selected from an FIR filter, an IIR filter and a self-adaptive hybrid filter according to different working conditions; the hydrogen production process is filtered, and the hydrogen charging and discharging amount of a hydrogen storage tank is adjusted according to the difference between the target hydrogen production rate and the actual hydrogen production rate, so that smooth hydrogen output is obtained; and evaluating the performance of the hydrogen production system according to the fluctuation ratio in the hydrogen output process, and readjusting the parameters of the filter according to the evaluation result. According to the invention, appropriate filter types are dynamically selected according to different working conditions, so that a relatively smooth hydrogen output rate is realized, the stability of the system is ensured, and the requirements of subsequent large-scale industrial production are met.

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 present application discloses an infinite pulse response filter and a method for implementing the filtering thereof. The IIR filter provided in the embodiment of the present application utilizes synchronous sampling technology to perform IIR calculation on multi-channel parallel input data in a time-sharing manner, thereby improving the IIR calculation speed, thereby reducing resource consumption, and greatly expanding the application scope of the IIR filter.

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 invention discloses a sensor data filtering method. The method comprises a data sampling stage; carrying out primary sampling on data output by an analog-to-digital converter in the sensor system, and storing the sampled data to a corresponding variable Cv; a monitoring variable calculation stage; calculating filtering results of short-time filtering, long-time filtering, super-long-time filtering and IIR filtering, and calculating filtering variations of the short-time filtering, the long-time filtering and the super-long-time filtering; a mode judgment stage; the filtering variable quantity obtained through calculation in the monitoring variable calculation stage is compared with a preset value to judge whether the sensor system enters a stable mode or a change mode; a data output stage; and in the stable mode and the change mode, outputting different filtering results respectively, and storing the output data to a variable Lv. The method can effectively reduce the interference of noise, monitors the slow change of the input data, and avoids the defect that the data trend is attenuated by times by conventional mean filtering.

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 radar detector employs parameterized subchannel analysis for discrimination of radar signals. Specifically, the signal processing of the radar detector includes a subchannel processor for evaluating programmable sub-bands of a channel for enhanced pattern identification. The subchannel processor specifically incorporates multiple digital demultiplexing stages, producing N signal subchannels at selectable frequencies. The subchannels are decimated from the original sampling frequency to a selectable lower sample frequency, e.g. using a cascaded integrator-comb filter, and then filtered, e.g. using a finite impulse response filter. The resulting sub channels 0 to N−1 are delivered for Neural Network assessment. Importantly, the subchannel frequencies, decimation rates and IIR filter profiles may be selectively adjusted by the control circuits to emphasize relevant patterns to be extracted by the neural network from a received radar signal.

The present disclosure provides methods, apparatuses, and systems for active noise control (ANC). The present implementations more particularly relate to filter adaptation techniques for recursively training an infinite impulse response (IIR) filter to convert a reference audio signal into an anti-noise signal. In some aspects, an ANC system may record a reference audio signal representing external noise via a feed-forward microphone, and may further record an error signal representing residual noise via a feedback microphone, the residual noise resulting from passive attenuation of the external noise via a main path between the microphones. The ANC system may further estimate an IIR filter that converts the reference audio signal to an error signal based on the secondary path. In some implementations, the ANC system may recursively update IIR filter coefficients based at least in part on coefficients associated with an error signal and a previous frame of a reference audio signal.

The invention discloses a data processing method applied to laser radar ranging, comprising: using an IIR filter to filter data with a bit width of W bits from a TDC; counting the first M bits of the W-bit data after filtering to obtain data T1 corresponding to a peak value, and counting BATCH data in total, wherein BATCH is a preset value; selecting data with the first M bits being equal to T1 in the data output by the TDC, counting the middle M bits of these data, obtaining data T2 corresponding to the peak value, and counting BATCH data in total; selecting data with the first 2M bits being equal to [T1, T2] or [T1, T2]-1 in the data output by the TDC, counting the last N bits of these data, obtaining data T3 corresponding to the peak value, and counting 3*BATCH data in total; obtaining a statistical result of [T1, T2, T3]; wherein W is the bit width of the TDC, M and N are preset values, M and N need to satisfy W=M+M+N, and BATCH is jointly determined according to the output frame rate of on-chip statistics and the frequency of TDC input data.

The utility model discloses an audio processor capable of presetting FIR filter sound box data, which comprises a box body, a signal transmission module, a DSP audio signal processing module, a control interface module and a power supply module, the signal transmission module is fixedly connected to the upper end of the box body, the control interface module is fixedly connected to the lower end of the box body, and the DSP audio signal processing module is fixedly connected to the lower end of the box body. The power supply module is fixedly connected to the right side in the box body. The audio processor further comprises an IIR filter module and an FIR filter module. The utility model belongs to the technical field of filter sound box data, and particularly relates to an audio processor capable of presetting FIR filter sound box data.

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.

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

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.

A method of monitoring a material yield sensor assembly for upward baseline drift, the method comprising steps of receiving a signal representing a plurality of raw baseline datapoints from the material yield sensor assembly, passing the raw baseline datapoints through an IIR filter thereby producing a plurality of filtered datapoints, fitting the filtered datapoints to a curve via a curve fitter thereby producing a plurality of fitted datapoints, and generating an alarm indicating the material yield sensor assembly should be serviced if a fitted datapoint of the plurality of fitted datapoints is greater than a first threshold.