12 results about "Iir filtering" patented technology

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 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.

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.

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 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.

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.

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.

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 present application relates to the technical field of power cable state monitoring and defect positioning, and discloses a partial discharge noise reduction method and system based on adaptive principal component analysis, comprising: constructing a noise feature database, using SVMD to adaptively decompose the partial discharge signal to be denoised, extracting intrinsic mode components through a constraint minimization model, and using a cross-correlation coefficient to identify and screen effective components; constructing a Gaussian mixture model and counting a priori, separating noise and partial discharge signals through a residual network and variational inference; combining a FIR filter and an IIR filter to perform deep noise suppression and signal reconstruction on the separated partial discharge signal; and using a closed-loop Bayesian calibration mechanism to dynamically optimize the reconstructed partial discharge signal, so as to obtain a denoised partial discharge signal. The present application can adaptively denoise the cable partial discharge signal, and improve noise suppression capability, signal identification accuracy and efficiency.

The application provides a sound source positioning method, system, device and program product. The method comprises: collecting multiple initial audio signals of a microphone array, the microphone array comprising multiple array-arranged microphones; performing filtering processing on each initial audio signal corresponding to each microphone by using a corresponding IIR filter and a corresponding all-pass phase compensation filter to obtain multiple filtered audio signals; wherein the IIR filter is configured to compensate for amplitude frequency response differences of the multiple microphones, and the all-pass phase compensation filter is configured to compensate for nonlinear phase distortion of the corresponding IIR filter; and determining sound source position information in a space where the microphone array is located based on the multiple filtered audio signals. The application solves the problem of frequency response differences caused by hardware differences of multiple microphones in a microphone array, and improves robustness.

The present disclosure relates to a filtering architecture with minimized transients. A digital microphone includes an analog-to-digital converter (ADC) and a digital filter system coupled to the ADC, where the digital filter system switches between a standard IIR filter architecture and a polyphase IIR filter architecture.