30 results about "Infinite impulse response" patented technology

The invention relates to the technical field of audio signal processing, and provides a self-adaptive frequency compensation system for small-sized multichannel sound equipment and a control method thereof, and the self-adaptive frequency compensation system for the small-sized multichannel sound equipment comprises an audio signal acquisition module, a frequency analysis module and a dynamic filtering compensation module which are electrically connected with one another in a communication manner; the audio signal acquisition module is used for acquiring an audio signal at a preset sampling rate through the ADC module; the frequency analysis module is used for executing FFT (Fast Fourier Transform) on the acquired audio signals to calculate energy distribution of corresponding frequency bands and identify missing or attenuated frequency ranges; and the dynamic filtering compensation module is used for selecting an FIR (Finite Impulse Response) or IIR (Infinite Impulse Response) filter according to a frequency analysis result, dynamically adjusting a filter coefficient and a gain value, performing real-time filtering processing on the audio signal and outputting a compensated audio signal. The missing frequency is identified and compensated by performing analysis, filtering, equalization and other processing on the audio signal.

This system for voice data processing comprises: an analog voice processing unit that receives an input of an analog voice signal and generates a digital voice signal; and a feature extraction unit that, when the digital voice signal is input, operates as an infinite impulse response (IIR) filter to filter the digital voice signal, and operate as an RNN to extract a voice feature from the filtered digital voice signal.

The invention relates to a digital filter circuit for calculating an exponential variance of a signal, a corresponding system on chip and a method of operation. A digital signal processing method includes: applying a first infinite impulse response filtering operation to a digital input signal to produce a first filtered signal; performing a mathematical transformation on a combination of the digital input signal and the first filtered signal to produce a transformed signal; applying a second infinite impulse response filtering operation to the transformed signal to produce a second filtered signal; performing a scaling operation on the second filtered signal to generate a scaled signal; and combining the second filtered signal and the scaled signal to produce a digital output signal indicative of a statistical characteristic of the digital input signal.

The apparatus and methods of the present invention provide improved accuracy of response for a tactile transducer included in a body-mounted device such as a headphone, VR / AR headset or similar device. Accuracy is increased through the application of digital signal processing, such as with Infinite Impulse Response filters or Finite Impulse Response filters.

The application discloses a decision feedback equalizer for PAM4 encoding band infinite impulse response filter, belongs to the field of signal processing, and comprises a high-pass filter and a linear equalizer, a decision feedback adder, and an infinite impulse response filter coefficient feedback control module; noise filtering is performed on a high-speed signal, low-frequency attenuation of the high-speed signal is compensated, a tailing effect in a frequency response is eliminated, high-speed inter-symbol interference is suppressed and eliminated, and an offline post-cursor in the frequency response is fed back and compensated; meanwhile, aiming at PAM4 encoded data, an RLM (Level separation mismatch ratio) calibration module and a clock data recovery circuit are added, which are used for improving the speed of clock receiving and recovery, and due to the addition of more high-speed modules, the load of the equalization adder is increased; the application uses a layout planning of a double-difference six-port mutual inductance bridge type (T-coil) inductive network, and bandwidth is expanded by using inductive peaking technology.

The invention provides a low-noise digital bandwidth compensation method for high-order frequency response of a sensor. The signal-to-noise ratio in the compensation process is remarkably improved while the signal bandwidth is expanded. The method comprises the following steps: S1, constructing a lumped parameter model based on a sensor actual measurement physical process, and deducing and calculating an overall high-order transfer function of the lumped parameter model; s2, acquiring actual sensor frequency response data and fitting the actual sensor frequency response data to determine model parameters in the high-order transfer function; s3, constructing an ideal expectation response model according to the target compensation bandwidth, and dividing the transfer function of the ideal expectation response model by the fitting transfer function obtained in the step S2 to obtain a continuous time transfer function of the compensation filter; s4, discretizing the continuous time transfer function of the compensation filter to obtain an infinite impulse response filter; and S5, cascading the infinite impulse response filter with a low-pass digital filter to form a final compensation filter to carry out bandwidth compensation on the output signal of the sensor.

Systems, methods, software, and devices are disclosed herein that transform a spatial input into a modal output that includes learned modal components of impulse responses. A neural network interpolates the modal components of impulse responses based on a desired sound source direction represented in the spatial input. The learned modal components are then used to determine coefficients of an infinite impulse response filter used to convert anechoic audio to spatialized audio. The spatialized audio provides a directional effect to a listener from the desired sound source direction.

The invention relates to the field of signal processing, in particular to a multi-lead signal filtering method and system based on an IIR (Infinite Impulse Response) high-pass filter. Acquiring a plurality of historical input signal sequences of the multi-lead signals in the previous time zone, and respectively performing signal prediction to obtain a plurality of predicted input signal sequences; randomly configuring a plurality of filtering coefficients, and calculating to obtain historical filtering cost and multi-lead filtering cost in combination with a plurality of historical filtering coefficients; performing correction calculation on the multi-lead filtering cost according to the plurality of prediction input signal sequences in combination with the prediction error coefficient to obtain a corrected multi-lead filtering cost; and according to the plurality of predicted input signal sequences and the plurality of filtering coefficients, carrying out filtering prediction, obtaining signal fitness, carrying out correction calculation on historical filtering cost, obtaining corrected historical filtering cost, carrying out filtering coefficient optimization, obtaining a plurality of optimal filtering coefficients, and carrying out filtering processing. The quality of multi-lead signal filtering is improved, and the reliability and accuracy of signals are improved.

Technologies of a device-based Fast Presence Detection (FPD) for a contactless sleep-tracking device are described. One method of a sleep-monitoring device includes receiving radar data from a radar unit. The radar data includes i) first data representing a breathing waveform associated with a user, ii) a first set of range values, and iii) a first set of confidence values associated with the first data. The method determines absolute magnitude values, first infinite impulse response (IIR) values using the first set of range values, and second IIR values using the first set of confidence values. The method determines a first event representing the user located in a first region using the absolute magnitude values and the first and second IIR values. The method sends an indication of the first event to a cloud service that causes one or more devices in the environment to perform one or more actions.

A computer-implemented method for mitigating noise from QRS complexes extracted from an electrocardiogram (“ECG”) signal includes: pre-processing an acquired ECG signal to obtain a pre-processed ECG signal identifying narrow pulses in the pre-processed ECG signal; extracting QRS complexes from the acquired ECG signal; cross-checking the extracted QRS complexes with the identified narrow pulses; and rejecting pulses in the extracted QRS complexes that synchronize with the identified narrow pulses to obtain denoised QRS complexes. The pre-processing includes applying a first bandpass filter comprised of finite and infinite impulse response filters; and applying a second bandpass filter comprised of an infinite impulse response filter.

A mode adaptive coefficient filter system for a physiological patent monitoring (“PPM”) method and apparatus provides an electrocardiogram (“ECG”) display function with an improved resolution of artifact issues when large baseline offsets occur to input signals. In one aspect of the disclosure the method and system provide an ECG monitoring and display function using the mode adaptive coefficient filter system for an active filter process provides for an improved ECG artifact resolution minimizing discontinuous wave forms when applied to an input signal that achieves improved signal quality and reduced latency in signal rendering. The method may be utilized for both infinite impulse response (“IIR”) and finite impulse response (“FIR”) systems with similar filter structures.

The invention discloses an error feedback noise shaping successive approximation analog-to-digital converter circuit, which comprises a successive approximation analog-to-digital converter, and a conversion residual voltage extraction amplification circuit and a switched capacitor infinite impulse response circuit which are electrically connected with the successive approximation analog-to-digital converter, one end of the conversion residual voltage extraction and amplification circuit is connected with the capacitor array digital-to-analog converter and is used for acquiring a conversion residual voltage signal, and the other end of the conversion residual voltage extraction and amplification circuit is connected with one end of the switched capacitor infinite impulse response circuit; and the amplified conversion residual voltage signal is fed back to the switched capacitor infinite impulse response circuit, and noise shaping is carried out through the noise shaping circuit. The invention discloses an error feedback noise shaping successive approximation analog-to-digital converter circuit with a simple structure and a conversion method thereof, which can realize error feedback noise shaping by utilizing a charge sharing summation mode, avoid the use of an operational amplifier and reduce power consumption.

A method of digital signal processing includes applying a first infinite impulse response filtering operation to a digital input signal to produce a first filtered signal, performing a mathematical transformation on a combination of the digital input signal and the first filtered signal to produce a transformed signal, applying a second infinite impulse response filtering operation to the transformed signal to produce a second filtered signal, performing a scaling operation on the second filtered signal to produce a scaled signal, and combining the second filtered signal and the scaled signal to produce a digital output signal indicative of a statistical property of the digital input signal.

The invention discloses a video metadata generation method, device and equipment and a readable storage medium, and the method comprises the steps: receiving a target video frame of to-be-generated metadata, and determining a previous video frame of the target video frame as a reference video frame; calculating a perception difference between the target video frame and the reference video frame; determining the redundancy of the target video frame according to the perception difference; determining initial metadata corresponding to the target video frame according to the redundancy; and carrying out infinite impulse response time domain filtering on the initial metadata to obtain target metadata. According to the method, the video metadata calculation burden is relieved, the calculation power overhead is reduced, the power consumption is reduced, and the cost is saved.

This invention discloses a disturbance compensation method based on a cyclic adaptive identification radial basis function neural network. The method includes: first, introducing an error transformation function and calculating the instantaneous quantity, then averaging it to obtain the average instantaneous quantity. Next, acquiring the q-axis current, first filtering it using an infinite impulse response filter, then filtering it a second time using a moving average filter, and constructing a trend curve of the q-axis current. Differentiating this trend curve using a backward differential, and then performing phase response compensation. Combining this compensation signal with the identification rate to generate the operating condition identification quantity. Finally, constructing a cyclic adaptive identification radial basis function neural network, using the average instantaneous quantity and the operating condition identification quantity as inputs, injecting the compensation quantity generated by the network into the current loop to achieve closed-loop compensation control of the system. By using this invention, efficient feedforward compensation for periodic disturbances is achieved. This invention can be widely applied in the field of permanent magnet synchronous motor control technology.

An infinite impulse response filter includes at least one N x N multimode optical coupler and one or more mode scrambling loops. The at least one N x N multimode optical coupler has N input ports and N output ports. One of the N input ports is coupled to a laser source that generates a light beam. The at least one N x N multimode optical coupler splits the light beam into a plurality of sub-beams that are partitioned in an amplitude domain across the N output ports. One of the N output ports is coupled to a detection system. The one or more mode scrambling loops include a mode scrambler and couple one of the N output ports to one of the N input ports. The one or more mode scrambling loops produce temporal incoherence and spatial incoherence that reduce peak power and speckle contrast of the light beam.

This invention discloses a method, apparatus, device, and medium for PPG spectrum enhancement based on adaptive weighting. The method includes: performing a Fast Fourier Transform on the PPG signal after removing motion artifacts to generate continuous spectral frame data, and calculating the signal-to-noise ratio (SNR) of the spectral frame data; quantizing the user's current motion level based on the motion acceleration component signal, and dynamically determining the fusion weight between the current spectral frame data and historical spectral frame data based on the calculated SNR of the current spectral frame data; fusing the current spectral frame data and historical spectral frame data using an infinite impulse response recursive weighted accumulation algorithm based on the determined fusion weight between the current spectral frame data and historical spectral frame data to obtain enhanced current spectral frame data; and detecting the user's current heart rate based on the enhanced current spectral frame data. This invention denoises the PPG signal with low computational overhead, improving the accuracy of user heart rate detection.

The invention provides a time-space synchronous acquisition method for video sound wave data, which belongs to the technical field of power transmission and transformation engineering, and comprises the following steps: synchronously acquiring video images and sound wave signals, and improving the sound wave measurement precision by adopting cross-correlation time delay estimation and IIR (Infinite Impulse Response) notch filtering; a double-layer game optimization model and a forward clustering method are utilized to realize heterogeneous data stream time alignment, non-uniform motion state parameters of a probe are estimated based on unscented Kalman filtering recursion, and a B-spline curve is adopted to fit a nonlinear mapping function of a depth position and time to correct spatial positioning deviation in a variable speed scene. Characteristic level semantic verification is achieved through a multi-mode space-time alignment mapping model, a synchronous alignment data set containing accurate space-time annotation is output, and the problem that due to non-uniform-speed motion of a probe, the depth position and time are in a nonlinear relation, and consequently video sound wave data space positioning error accumulation is caused is solved.

Apparatus and associated methods relate to global implied volatility assessment in a dynamic inertia system. In an illustrative example, a global implied volatility assessment system (GIVAS) may include a market data standardization module configured to receive updates of option contracts of a cryptocurrency from multiple data tracking devices. For example, the option contracts value may be prone to outlying events causing discontinuity in a time-series of the value. The received update may, for example, be aggregated into a global order book (GOB) including instantaneous representations of the option contracts among the multiple data tracking devices. Based on the GOB, the GIVAS may generate a global raw volatility characterization (GRVC) of the option contracts. An infinite impulse response filter may be applied to the GRVC to generate a transient-dampened volatility characterization. Various embodiments may advantageously generate a transient-dampened volatility metric usable for analyzing the option contract value by external code.

The invention discloses a decision feedback equalizer for a PAM4 coding band infinite impulse response filter, which belongs to the field of signal processing and comprises a high-pass filter, a linear equalizer, a decision feedback adder and an infinite impulse response filter coefficient feedback control module. Noise filtering is carried out on the high-speed signals, low-frequency attenuation of the high-speed signals is compensated, the trailing effect in frequency response is eliminated, high-speed inter-symbol interference is suppressed and eliminated, and offline Post-cursor in the frequency response is fed back and compensated; meanwhile, aiming at PAM4 coded data, an RLM (Level Segment Mismatch Rate) calibration module and a clock data recovery circuit are added to improve the clock receiving and recovering speed, and due to the addition of more high-speed modules, the load of the balance adder is increased, the layout planning of a double-difference six-port mutual inductance bridge type (T-coil) inductance network is applied, and the balance adder has the advantages that the structure is simple, the cost is low, and the cost is low. And the bandwidth is expanded by using an inductance peaking technology.

The present disclosure relates to clock synchronization using periodic external reference calibration. According to one embodiment, a touch controller includes an oscillator that generates a low frequency reference signal, a phase locked loop that generates a high frequency clock signal based on the reference signal, and a calibration circuit that measures a timing difference between the high frequency clock and a periodically enabled external reference clock from a processor. The calibration circuit performs multiple measurements during an initial enablement period and averages the validated measurements to establish a baseline timing parameter. During subsequent enablement periods, new measurements are filtered with previous values using an infinite impulse response filter. A clock divider generates a timing signal using a nominal ratio and periodically adjusts between the nominal ratio and a modified ratio based on the filtered measurements. The touch controller performs capacitance measurements on touch panel electrodes, transmits uplink signals through voltage modulation, and demodulates downlink signals during defined time slots.

The invention relates to the technical field of integrated circuits, and provides a digital filtering device, a chip and a control method thereof, and the device comprises a first delay module which is used for latching k delay input signals; the second delay module is used for latching k + 1 delay output signals; the input end of the first multipath selection module is connected with the output end of the first delay module and the output end of the second delay module; the second multi-path selection module is used for acquiring a multiplication coefficient; the counting module is used for selecting corresponding input data and multiplication coefficients in a time-sharing manner; the multiplication module is used for generating a multiplication result; the third delay module is used for latching a preorder accumulation result; the input end of the addition module is connected to the output end of the multiplication module, and the addition module is used for adding the current multiplication result and the preorder accumulated value; the shifting module is used for obtaining a scaling result; and the first rounding module is used for performing bit width adjustment on the scaling result to obtain a current output signal. The device is used for carrying out Butterworth infinite impulse response filtering.

Inductive or magnetic sensor (1) with at least one sensor element (2) for detecting sensor signals and at least one control and evaluation unit (3) for generating application parameters from the sensor signals, and an output unit (4) for outputting at least one object detection signal, wherein the control and evaluation unit (3) has at least one filter (5) for filtering the application parameters and generating and outputting filter results, where the filter (5) is a filter with an infinite impulse response, the filter (5) is an adaptive filter, and the control and evaluation unit (3) has the filter (5) and is configured to form the object detection signal depending on the filter results of the filter (5), characterized in that the filter (5) has at least four filter parameters, wherein the filter parameters at least one filter value as the starting value of the filter (5), at least one filter value as the final value of the filter (5), at least one incremental value of the filter value and at least one filter counter with a number of values ​​to be determined within a filter stage are, wherein the filter (5) is designed to cyclically capture the application parameters at different times according to the number of values ​​to be determined, and to filter with the filter value as the starting value of the filter (5), and to calculate an updated filter result, wherein the application parameters are cyclically acquired by the filter (5) at different times according to the number of values ​​to be determined, and are filtered with the current filter value, wherein the application parameters are cyclically recorded by the filter (5) at different times according to the number of values ​​to be determined and are filtered with the most recent filter value until a final value of the filter (5) is reached, wherein the control and evaluation unit (3) is designed to evaluate the final filtered filter result in the control and evaluation unit (3) for the calculation of the object detection signal.

Filter method for converting an analog input signal into a sampled digital output signal, wherein the z-transformed digital output signal Y(z) results from the z-transformed analog input signal X(z) and the z-transformed rounding error E(z) at order k as Y(z) = X(z) · z -1 + E(z) · (1- z -1 ) k , where Y r (z) = E(z) · (1-z -1 ) k the z-transform of the noise signal is characterized in that - the filter gain at the frequency limit approaches 0 to 1, - for an impulse response h i of the filter at time i for a filter length N: ∑ i = 0 N − 1 h i = 1, where the filter is set up such that the ratio of the variance of the noise signal is var(Y r ) to the variance of the rounding error var(E) yields: var ( Y r ) var ( E ) = h 0 2 + ( h 1 − h 0 ) 2 + ( h 2 − h 1 ) 2 + ⋯ + ( h N − 2 − h N − 1 ) 2 + h N − 1 2 and the variance of the noise signal var(Y r ) is minimized, where the filtering method is an infinite impulse response method.

The invention relates to the technical field of communication, and discloses an ultra-wideband signal compensation method and device, equipment and a storage medium. The method comprises the following steps: acquiring an ultra-wideband signal to be compensated, and acquiring an amplitude-frequency response expression of an analog filter according to the ultra-wideband signal; obtaining a denominator coefficient of the analog filter according to the ultra-wideband signal through a fitting method, and obtaining a denominator coefficient and a numerator constant of the transfer function according to the amplitude-frequency response expression and the denominator coefficient of the analog filter; and according to the denominator coefficient and the numerator constant of the transfer function, generating a target infinite impulse response filter, and realizing flatness compensation of the ultra-wideband signal based on the target infinite impulse response filter. According to the scheme of the embodiment, the filter parameters are obtained based on the fitting method, and the target infinite impulse response filter is generated based on the filter parameters, so that flatness compensation is realized, effective flatness compensation can be carried out on the ultra-wideband signal, and the compensation cost can be reduced at the same time.

According to an embodiment, a regression analysis is performed on a subset of a dataset, where the subset of the dataset corresponds to inputs from a first row of a matrix of sensors at a time instant k. The regression analysis generates a set of coefficients. A filter transform, to be applied on the subset of the dataset, is determined based on a comparison between the set of coefficients and threshold values. The filter transform can be one of an infinite impulse response (IIR) filter transform, a first-order filter transform, or a second-order filter transform. Once the filter transform is determined, it is applied to the subset of the dataset to generate a first output matrix.

The invention relates to the technical field of deep learning and neural network model compression, and discloses a model filter compression method based on gradient guidance and terminal equipment in order to solve the technical problems of high reasoning cost and difficult deployment of an FSMN model caused by a high-order FIR filter. A trained FSMN model is obtained, the model comprises at least one high-order finite impulse response filter layer, the weight of the high-order finite impulse response filter layer is defined as the step b, and the gradient of a final loss function of the FSMN model relative to the gradient is determined and calculated; and step c, searching and determining an infinite impulse response filter of which the order is lower than that of the FIR filter based on the guidance of the gradient, and defining the weight of the infinite impulse response filter as step d, and replacing the generated compressed FSMN model with the infinite impulse response filter. The problem that the FSMN model is high in reasoning cost is solved, and the parameter quantity and the calculation complexity of the model are remarkably reduced.

The invention discloses a planned sampling non-autoregression learning method for acoustic feedback suppression, which belongs to the technical field of audio signal processing and deep learning, and comprises the following steps: constructing a non-autoregression open-loop training framework; constructing a two-stage limited boundary plan sampling mechanism; dynamically selecting an input source of a current frame between teacher forced input and a model prediction result according to a preset probability scheduling strategy; constructing a finite-order Newman series approximation operator, performing approximation modeling on an infinite impulse response recursive structure of acoustic feedback, and generating feedback estimation characteristics consistent with closed-loop operation behaviors; and performing staged model training to enable the model to gradually adapt to closed-loop reasoning conditions. According to the method, by introducing planned sampling and finite order Newman series approximation, the model can adaptively learn an internal mode of feedback signals evolved along with time, so that the stable performance is kept under different system gains, acoustic coupling intensities and operation scenes, and the method has good engineering adaptability and application prospects.

The application discloses a bias compensation adaptive filtering method based on a matrix eigenvalue decomposition method and belongs to the field of digital filters.The application is implemented in the following manner: a corresponding bias compensation method is selected according to a filter type, and a corresponding bias compensation adaptive filtering step is executed; the filter type is divided into a finite impulse response filter and an infinite impulse response filter, wherein the infinite impulse response filter includes two cases of white output noise and colored output noise.The application adopts the eigenvalue decomposition of a matrix to obtain unknown parameters of a system, and realizes unbiased estimation of system parameters of a bias compensation adaptive filter; in the calculation process, the application does not need to estimate the variance of output noise, does not need to derive a cost function multiple times, and only needs to estimate the variance of input noise, thereby eliminating the influence of the variance of output noise on an unknown parameter estimated value and reducing the amount of operation parameters.The application has the advantages of high estimation precision, good filtering effect, good robustness and wide application range.