65 results about "Linear filter" patented technology

The invention provides an OBD message analysis and attitude rendering method and system based on a vehicle dynamics model, and belongs to the field of vehicle electronics and control. The method comprises the steps that a message is analyzed through an OBD interface, and vehicle operation parameters are obtained; establishing a unified time base and carrying out resampling and synchronous processing on the data of each channel; constructing a fifteen-degree-of-freedom vehicle dynamics model comprising a vehicle body six-degree-of-freedom pose, a vehicle body angular velocity, a four-wheel rotation angular velocity and unsprung mass vertical displacement for state prediction; designing an observation function to map a model state to an OBD observation space, and adopting an observability enhancement strategy; predicting and observing data are fused by using a nonlinear filter, and abnormal observation gating is carried out based on a mahalanobis distance and an amplitude threshold value; compensating system time delay and adaptively adjusting observation noise; and performing three-dimensional real-time rendering on the attitude information. According to the method, high-precision and high-robustness vehicle attitude estimation can be realized only by depending on the OBD message, and the system cost is effectively reduced.

In various examples, the present disclosure relates to using temporal filters for automated real-time classification. The technology described herein improves the performance of a multiclass classifier that may be used to classify a temporal sequence of input signals—such as input signals representative of video frames. A performance improvement may be achieved, at least in part, by applying a temporal filter to an output of the multiclass classifier. For example, the temporal filter may leverage classifications associated with preceding input signals to improve the final classification given to a subsequent signal. In some embodiments, the temporal filter may also use data from a confusion matrix to correct for the probable occurrence of certain types of classification errors. The temporal filter may be a linear filter, a nonlinear filter, an adaptive filter, and / or a statistical filter.

Systems and methods for compensating a transmit signal for charge trapping effects of a power amplifier are provided. In certain embodiments, a non-linear filter is trained based on time aligning a first set of observations taken from digital transmit data prior to conversion to a radio frequency transmit signal, and a second set of observations taken from an output of a power amplifier that amplifies the radio frequency transmit signal. In certain implementations, the first set of observations and the second set of observations are obtained without decimation. Rather, decimation is provided after timing alignment. By implementing the DPD system in this manner, signal data is not lost by decimation and more accurate timing alignment between the sets of observations is achieved.

A noise reduction system for actively compensating background noise in a passenger transport area of a vehicle. The noise reduction system includes a nonlinearity filter unit having a model of a non-linear transfer function of the sound generator, wherein the nonlinearity filter unit is configured to receive the anti-noise signal and to generate a filtered anti-noise signal by applying a non-linear filter function on the anti-noise signal, which is based on the model of the non-linear transfer function in that the non-linear response of the sound generator is at least partially corrected when driven by the filtered anti-noise signal. Wherein the nonlinearity filter unit is further configured to output the filtered anti-noise signal to a sound generator.

A memory device is operated using a reduced row address space (e.g., 48k rows per bank vs. a full capacity 64k rows per bank). The unused address space (e.g., 16k rows) is used and mapped by the memory device to store information for at least some of the bad rows. In this manner, many more bad rows may be "repaired" at the expense of a reduction in the capacity of the memory device. The translations of external row addresses to internal physical row addresses may include permuting row addresses within groups of row addresses to avoid bad rows within respective groups of row addresses. A linear filter (e.g., Bloom Filter, XOR filter, etc.) may be used to store information for the translations of external row addresses to internal physical row addresses. The information for the translations may be determined and programmed into the memory device once during manufacturing.

The present invention provides an echo cancellation method in a delay jitter scenario in the field of echo processing technology, comprising: step S10, collecting a far-end signal and a near-end signal; step S20, calibrating and aligning the far-end signal and the near-end signal through a delay calibration module and then inputting them into a linear filtering module; step S30, the linear filter of the linear filtering module outputs a linear filter coefficient and an error signal to a far-end signal synthesis unit based on the far-end signal and the near-end signal; step S40, the far-end signal synthesis unit calculates a far-end reference signal and a delay state based on the linear filter coefficient, and outputs the far-end reference signal, error signal and delay state to a nonlinear suppression module; step S50, the nonlinear suppression module outputs an echo-free signal based on the far-end reference signal, error signal and delay state. The advantages of the present invention are: it realizes compensation for the delay jitter of the echo, greatly improves the echo cancellation effect, and thus greatly improves the call quality.

The application discloses a linear filtering algorithm optimization method based on a depth separable convolution idea, and comprises the following steps: S1, generating the weight of a filter kernel according to a selected linear filtering algorithm and a set filter kernel size; S2, filling the image boundary according to the set filter kernel size; S3, processing an input image according to the channel number of the input image; S4, merging the multi-channels after the linear filtering operation, and outputting the filtered image; S5, using mean square error, peak signal-to-noise ratio and structural similarity as indexes to analyze the image quality of the output filtered image; and S6, comparing the linear filtering algorithm with a naive linear filtering algorithm, a linear filtering algorithm based on a row-column separation idea and a linear filtering algorithm of an OpenCV library based on a buffer and the row-column separation idea. The application can effectively adapt to various hardware environments without the need of making a large number of modifications for specific chips.

A method of operation of a texturing / shading unit in a GPU pipeline is used for efficient convolution operations. The method uses texture hardware to collectively fetch all the texels required to calculate properties for a group of output pixels without any duplication. The method then bypasses bilinear filter hardware in the texture hardware and passes the fetched and unfiltered texel data from the texture hardware unit to shader hardware in the texturing / shading unit. The shader hardware uses the fetched texel data to perform a plurality of convolution operations to calculate the properties of each of the output pixel.

The application relates to an audio signal enhancement method and device, computer equipment and a storage medium. The method comprises the following steps: when a voice packet is received, sequentially decoding and filtering the voice packet to obtain an audio signal; when the audio signal is a forward error correction frame signal, extracting a feature parameter from the audio signal; converting the audio signal into a filter speech excitation signal based on a linear filtering parameter obtained by decoding the voice packet; performing speech enhancement processing on the filter speech excitation signal according to the feature parameter, a long-time filtering parameter and a linear filtering parameter obtained by decoding the voice packet, to obtain an enhanced speech excitation signal; and performing speech synthesis based on the enhanced speech excitation signal and the linear filtering parameter to obtain a speech enhancement signal. The method can improve the timeliness of audio signal enhancement.

The application provides a method and device for intra-channel and inter-channel nonlinear joint compensation of a WDM system based on an improved LDBP and electronic equipment, and belongs to the technical field of optical fiber communication. The method compensates dispersion by using a linear filter in the time domain, the corresponding network structure is designed as one-dimensional convolution, and the pulse broadening effect is combined in the overlap reservation method, the physical model of nonlinear compensation is improved by considering the interference of intra-channel dispersion pulse broadening and the walk-off effect between channels, and the purpose of intra-channel and inter-channel nonlinear joint compensation is achieved. The application fully utilizes the random interaction between inherent dispersion and nonlinearity in the optical fiber communication system to compensate the phase shift caused by the nonlinear effect, can effectively equalize the nonlinear distortion of the signal under the condition of controlling the calculation complexity and reducing the implementation cost, and is expected to play a better role in the long-distance transmission of the large-capacity WDM system.

The invention discloses a tobacco online near infrared spectrum detection device and method based on a linear optical filter, and relates to the technical field of spectrum processing. Near-infrared light is emitted to a target area, a plurality of near-infrared bands are obtained through a linear optical filter, and reflected light of each near-infrared band is collected to obtain an original near-infrared band set; preprocessing the target original near-infrared band to obtain a target near-infrared band; performing feature extraction on the target near-infrared band to obtain target sub-features; and constructing a target feature vector according to all the target sub-features, and searching a preset database according to the target feature vector to obtain a detection result. Multiple near-infrared bands and frequency-combining and frequency-doubling absorption information of substance molecules corresponding to different bands are obtained through the linear optical filter, and then the different near-infrared bands are preprocessed and then subjected to feature extraction, so that prediction of different chemical substance components in tobacco is improved, and the online detection efficiency of the tobacco is improved.

This signal processing device comprises: a plurality of linear filters that remove interference and that include a first linear filter to which a reception signal and an auxiliary first input signal are inputted and which outputs an output signal and an auxiliary first output signal, and a second linear filter to which an auxiliary second input signal is inputted and which outputs an auxiliary second output signal; a signal equalization unit having a nonlinear signal conversion unit that converts a signal to be converted, in accordance with the modulation scheme of a transmission signal corresponding to the reception signal, and a phase shift unit that shifts the phase of a signal to be shifted; and a phase noise removal unit that estimates a phase fluctuation relating to the output signal. The phase shift unit shifts the phase of the signal to be shifted, in accordance with the estimated phase fluctuation, the auxiliary first output signal outputted by the first linear filter is inputted to the nonlinear signal conversion unit via the phase shift unit, the signal outputted by the nonlinear signal conversion unit is inputted as the auxiliary second input signal to the second linear filter via the phase shift unit, and the auxiliary second output signal outputted by the second linear filter is inputted as the auxiliary first input signal to the first linear filter.

Techniques are provided for non-linear filtering of a signal for improved pulse detection and pulse width discrimination. A system implementing the techniques according to an embodiment includes a first linear filter configured to filter an in-phase component of a received signal to a downsample bandwidth and a second linear filter configured to filter a quadrature phase component of the received signal to the downsample bandwidth. The system also includes a magnitude calculation circuit coupled to outputs of the first linear filter and the second linear filter and configured to generate a magnitude signal based on the filtered in-phase component and the filtered quadrature phase component of the received signal. The system further includes a median filter processor coupled to an output of the magnitude calculation circuit and configured to apply a median filter to the magnitude signal to generate a filtered signal having reduced noise while maintaining sharp edge transitions.

The invention provides an organ quality evaluation system and method based on visible-near infrared hyperspectral imaging, and the system employs a continuous gradual change optical filter which can achieve the continuous filtering function from visible light to near infrared light from 400 nm to 1000 nm, and a drive module to form a linear filtering control module, and is used for achieving the filtering of reflected light of a target object (organ) in different wave bands. A reflection spectrum of biological tissues such as hemoglobin and fat is obtained, and molecular fingerprint level data is provided for organ quality evaluation; the imaging camera and the upper computer control module form an imaging processing module, image data of isolated organs of different wave bands are collected through the imaging camera, and spectrum correction, image reconstruction, organ quality evaluation and result display are completed through the upper computer control module. According to the invention, rapid and non-destructive evaluation of the transplanted organ can be realized.

The invention particularly relates to a sea clutter pulse interference suppression method for cascaded Hammerstein robust filtering, and the method comprises the steps: receiving an echo signal, and carrying out the preprocessing of the echo signal, so as to obtain a direct signal and a sea clutter signal, which are separated from each other; a cascaded Hammerstein filter is used to carry out multi-stage filtering processing on the separation signal, and a non-linear mapping output signal is obtained; wherein the cascaded Hammerstein filter comprises a first nonlinear filtering module, a second linear filtering module and a third nonlinear filtering module which are arranged in a cascaded manner; and carrying out filtering processing on the nonlinear mapping output signal through a robust filter to obtain an output signal after interference suppression. According to the method, nonlinear distortion and non-Gaussian pulse interference can be effectively suppressed in a complex marine environment, and the anti-interference capability, the signal fidelity and the adaptivity of a radar and a communication system are improved.

Multiple successively spatially downsampled versions of a rendered image frame are generated for at least one two-dimensional signal component of one or more two-dimensional signal components of an im

The invention discloses a code hopping spread spectrum communication timing synchronization method and system, and the method employs a signal detection and frame synchronization method to carry out timing coarse synchronization, can calculate the noise floor in real time according to the change of an electromagnetic environment, takes the ratio of a useful signal incoherent processing result to a noise signal incoherent processing result as a detection quantity, is not sensitive to the change of the electromagnetic environment, and is high in detection precision. The risk of signal false capture or missing capture can be effectively avoided; a timing fine synchronization method is adopted, sampling errors are estimated through a correlation peak fitting mode, real-time correction of the sampling errors is achieved through an online table look-up and time domain linear filtering method, the method has the advantages of being high in estimation precision and good in real-time performance under the low signal-to-noise ratio, meanwhile, sampling error estimation and correction are conducted in the time domain, hardware resource overhead is effectively reduced, and the method is suitable for popularization and application. The antenna can meet the requirements of long-distance and high-reliability broadband communication under low power, and can be applied to the working scenes of short-time burst communication and continuous wave communication.

The application discloses a kind of echo cancellation processing method and device under the big scene of loudspeaker loudness, belong to the field of speech signal processing;The echo cancellation processing device is installed in equipment inside, including audio acquisition module, adaptive threshold compression module, delay determination module, delay compensation module, echo cancellation module and audio playback module;The echo cancellation method refers to: when equipment is online, audio acquisition module acquires the reference signal sent by remote device and near-end microphone signal, input to adaptive threshold compression module and carry out compression to the amplitude of microphone signal.Meanwhile, according to the delay τ stored in advance, delay compensation module is carried out to the delay compensation of remote reference signal;Finally, through echo cancellation module, the signal after compression limiting processing and the remote reference signal after compensation delay, carry out block segment frequency domain adaptive filtering, and carry out echo cancellation.The application makes that linear filter convergence speed is faster and more stable.

The present application relates to a kind of back wave field synthesis processing method, device, computer equipment and storage medium, method includes the following steps: respectively obtaining source back wave data and source wavelet data;According to source back wave data and source wavelet data, determine sparse linear filter coefficient;Using the numerical value of non-zero sparse linear filter coefficient and delay, time delay weighted superposition of forward wave field, synthesis back wave field.The above-mentioned back wave field synthesis processing method, based on source back data is the convolution of source wavelet data and sparse linear filter, determine sparse linear filter coefficient by source back data and source wavelet data, then according to the principle of superposition, time delay weighted superposition of forward wave field, synthesis back wave field.No need to rely on back data to obtain back wave field by solving wave equation, power is greatly reduced, can improve processing efficiency, reduce processing time, can also reduce memory consumption, so as to reduce power cost.

This disclosure relates to the field of seismic imaging technology, and particularly to a method, apparatus, device, and medium for generating common imaging point gathers for wave equations. The method includes: identifying a source wavelet based on a selected source location; synthesizing the return wavefield of a single-channel seismic record corresponding to a geophone based on the forward propagation wavefield of the source wavelet and a pre-determined linear filter; synthesizing a single-channel seismic image corresponding to the single-channel seismic record using the return wavefield and the forward propagation wavefield; calculating the offset distance corresponding to each single-channel seismic image; classifying the single-channel seismic images according to the offset distance; and superimposing the classified single-channel seismic images to obtain an offset common imaging point gather; performing amplitude processing on the offset common imaging point gathers; and determining the amplitude-processed offset imaging point gathers as the common imaging point gathers for wave equations. This disclosure can improve the efficiency of common imaging point gather generation.

The application discloses a remote sensing image target detection method based on an RFTZN neural dynamics model, and belongs to the technical field of remote sensing image target detection, which comprises the following steps: based on a to-be-detected remote sensing image and a detection target feature, a target detection task of the remote sensing image is constructed as a CEM optimization problem, and the CEM optimization problem is converted into an equivalent linear matrix equation; an RFTZN neural dynamics model is constructed; based on the RFTZN neural dynamics model, a solver is constructed, and the linear matrix equation is solved to obtain a linear filter; and the linear filter is applied to the to-be-detected remote sensing image to extract a detection target. The RFTZN neural dynamics model is constructed by innovatively introducing a feedback regulation mechanism and a dynamic self-adaptive scale factor, so that the solving speed and the anti-noise interference capability of the RFTZN neural dynamics model are far superior to those of a traditional model, and the dynamic optimization problem in a target detection process can be efficiently solved.

The embodiment of the invention provides a satellite receiver signal processing method and device, and the method comprises the steps: carrying out the demodulation processing of a received signal, and obtaining a baseband signal; performing correlation calculation processing on the baseband signal and a local correlation sequence to obtain a correlation value; constructing a signal space based on the correlation value; performing linear filtering processing on the correlation value of the signal space to obtain a linearly filtered signal space; based on the signal space after linear filtering, the maximum correlation value under the same Doppler frequency and different code phase delays in the first dimension is searched, the sum of the distances between the maximum correlation value and the correlation values under the same Doppler frequency and other code phase delays is calculated, and the maximum distance is selected from the sum of the distances corresponding to the Doppler frequencies; performing nonlinear filtering processing on the correlation value of the Doppler frequency corresponding to the maximum distance to obtain a correlation value after nonlinear filtering; and determining whether a target signal exists based on the correlation value after nonlinear filtering. The performance of capturing the target signal can be improved.

The application discloses a multi-frame filtering amplitude-phase decoupling deep neural network stereo echo cancellation method and system based on human ear hearing. The method comprises the following steps: using a microphone to receive a signal and two-channel reference signal to estimate the real part and the imaginary part of a linear filter, and using a multi-frame filter structure to estimate a linear echo signal in real time to obtain a residual echo complex spectrum; estimating a near-end speech amplitude spectrum, training a neural network for estimating the near-end speech amplitude spectrum through a human ear hearing related cost function; performing secondary estimation on the near-end speech complex spectrum, and training the neural network through the human ear hearing related cost function; and performing stereo echo cancellation through the trained neural network. The application decouples the mapping of the near-end speech complex spectrum in the stereo echo cancellation system into amplitude spectrum mapping and phase mapping, and finely recovers the near-end speech complex spectrum through secondary estimation, thereby improving the quality and intelligibility of the near-end speech.

The invention provides an IMU gyroscope optimization method and system based on a neural network, and relates to the technical field of IMU gyroscope optimization. The method comprises the following steps: firstly, acquiring a time sequence signal of the gyroscope, and processing the time sequence signal by utilizing linear filters with different bandwidths to obtain a multi-resolution signal branch; then, inputting each signal branch into an adaptive expansion convolutional network, generating a feature representation set, and calculating a cross-resolution attention weight; next, fusing the branch features by using the weights to obtain a fused feature set, inputting the fused feature set into a neural network to be optimized, and outputting the denoised angular velocity, bias estimation and attitude increment; and finally, constructing a multiple loss function based on the optimization parameters to carry out network training, and obtaining optimized angular velocity and attitude estimation by using actual measurement signals. According to the method, the problem that an IMU output signal is unstable due to the fact that attitude errors caused by time-varying bias and noise are difficult to effectively suppress in the prior art is solved.

A coder (encoder or decoder) determines, for a current block (CB), a reference block (RB) based on a template matching (TM) cost associated with a first reference template of the RB and a first current template of the CB. The coder further determines, based on the first reference template matching the first current template being flipped in a direction, coefficients of a linear spatial filter based on filtered values, resulting from the linear spatial filter applied to first samples of a second reference template of the RB, being set to corresponding values of second samples of a second current template flipped in the direction. An adjusted RB is generated by applying the linear spatial filter with the determined coefficients to the RB. The coder codes the CB based on a residual of the CB determined based on the adjusted RB.

Devices, systems and methods for temporal prediction of parameters in non-linear adaptive loop filtering are described. In an exemplary aspect, a method for visual media processing includes configuring, for a current video block, one or more parameters of a clipping operation that is part of a non-linear filtering operation; and performing, based on the one or more parameters, a conversion between the current video block and a bitstream representation of the current video block, wherein the one or more parameters is coded in accordance with a rule.

The application discloses a sample structure-based positioning table data splicing method and device, equipment and medium, and relates to the technical field of image processing. The method comprises the following steps: determining the splicing seam position data in the to-be-spliced positioning table, performing image rendering on the splicing seam position data by using bilinear interpolation to obtain a plurality of rendered images; integrating the pixel value size of each pixel in the plurality of rendered images based on linear filter transformation to obtain a plurality of continuity-enhanced images; determining a target pixel meeting a preset enhancement condition by using a preset linear functional, and enhancing the target pixel to obtain a plurality of target-enhanced images; determining an image registration offset, splicing the to-be-spliced positioning table according to the image registration offset to obtain a target spliced positioning table, then rendering the target spliced positioning table, and verifying the splicing effect of the obtained target spliced image. Thus, the splicing dislocation problem caused by the error of image registration can be reduced.

Aspects of the disclosure includes methods and apparatuses for video decoding and video encoding and a method of processing visual media data. The apparatus for video decoding includes processing circuitry configured to: receive coded information indicating that a current block in a current picture is predicted using a filtered intra block copy (FIBC) mode; determine a linear predicted value of a current sample in the current block by applying a linear filter to prediction samples predicted using one of an IBC mode and an intra template matching (IntraTMP) mode; determine a gradient value associated with the current sample using at least one gradient filter; determine a predicted value of the current sample based on a sum of the linear predicted value and at least one modification value that includes the gradient value. An FIBC filter in the FIBC mode includes the linear filter and the at least one gradient filter.

This invention discloses a broadband linear graded filter for spectral imaging, possessing high transmittance and excellent stray light suppression performance. The filter employs a multilayer thin-film structure, including an interference cutoff film system above a substrate and a narrowband filter film system below. The narrowband filter film system achieves high transmittance in the 500-800 nm wavelength range through an optimized design using multiple alternating combinations of high and low refractive index materials. The interference cutoff film system effectively suppresses stray light through its long-pass and short-pass design.