102 results about "Linear filter" patented technology

The invention discloses a coordinate system transformation fusion filtering tracking method and system for a dual-base-station radar, and relates to the technical field of radar target tracking. The method comprises the following steps: firstly, establishing a state and observation equation, and determining an initial state of a target based on prior information under Cartesian coordinates; then, obtaining a target motion state through one-step prediction, generating a sigma point by utilizing U transformation, and calculating a mean value and a covariance for constructing state space prediction; thirdly, fusing the measurement data and the prediction data by using a Kalman filter to obtain optimal state estimation and a covariance matrix; and finally, converting the updated state estimation back to the Cartesian coordinate system based on U transformation, and circulating the process until the tracking is finished. According to the method, the nonlinear filtering problem in the updating process can be avoided, and the tracking robustness and precision are improved.

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.

Methods for video decoding and encoding, apparatuses and non-transitory computer-readable storage media thereof are provided. In one method for video decoding, a decoder may obtain a plurality of first motion compensated samples of a first channel and a plurality of second motion compensated samples of a second channel for a current inter coding block, obtain a non-linear filter, and obtain a plurality of filtered motion compensated samples of the first channel based on the non-linear filter, the first motion compensated samples of the first channel, and the second motion compensated samples of the second channel.

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.

The invention discloses an audio authenticity identification method and device based on a dual-path Transform, and the method comprises the following steps: preprocessing: converting an original audio signal into time-frequency representation, and extracting features through LFCC or STFT to obtain the time-domain features and frequency-domain features of the audio signal; the time-frequency representation feature = {B, F, T} containing time domain features and frequency domain features is obtained through preprocessing, B is the batch size, F is the number of frequency channels equal to the number of LFCC linear filters, and T is the number of time frames; the convolution encoder is used for encoding the input time-frequency representation through a plurality of 2D convolution layers and extracting local features; and respectively processing the frequency domain information and the time domain information through a double-path Transform structure to form semantic-acoustic information decoupling. Frequency domain information and time domain information are respectively processed through a double-path Transform structure, and efficient decoupling of semantic information and acoustic information is realized in combination with an attention mechanism, so that audio information of different levels is better extracted and utilized.

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.

The invention relates to a clutter signal generation method and device. The method comprises the following steps: generating two groups of uniformly distributed random sequences u1 and u2; converting the two groups of uniformly distributed random sequences into a Gaussian white noise sequence through a Box-Muller algorithm; generating a Gaussian color noise sequence from the Gaussian white noise sequence through a linear filter; performing nonlinear transformation on the Gaussian color noise sequence to obtain a clutter basic model sequence; and performing amplitude modulation, distance modulation and speed modulation on the clutter basic model sequence to generate a clutter signal with amplitude, time domain and frequency domain information. According to the invention, amplitude modulation, distance modulation and speed modulation are respectively carried out on the clutter basic model sequence by using the clutter sequence modulation module, so that the generated clutter signal has amplitude, time domain and frequency domain information coupled with a simulation scene, and the accuracy and fidelity of clutter simulation are improved.

The invention belongs to the technical field of image processing, and discloses a single-source domain generalization medical image segmentation method and related device based on domain randomization, and the method comprises the steps: inputting a to-be-segmented medical image into a pre-trained segmentation network model, and obtaining a segmentation result of the to-be-segmented medical image; the segmentation network model comprises a global intensity nonlinear filter, an encoder and a decoder; a uniform low-frequency spectrum transformation filter is arranged in the encoder and is used for performing mixing operation on low-frequency components of the spectrum of the input image by using Bernoulli distribution according to statistical magnitude sampled from uniform distribution to obtain a disturbance low-frequency component and combining the disturbance low-frequency component with high-frequency components of the spectrum of the input image to obtain an augmented spectrum; and the decoder is used for fusing the input image, the feature map of the input image, which passes through the downsampling of the convolutional layer, and the multi-resolution feature map from the encoder through step-by-step upsampling, and generating a pixel-by-pixel segmentation result of the input image through the convolutional layer and an activation function, thereby improving the image segmentation capability of an unknown domain.

Methods for video decoding and encoding, apparatuses, and non-transitory computer-readable storage media are provided. In one method, a decoder may at least one block vector (BV) based on at least one BV candidate list for an intra template matching prediction (IntraTMP) mode. Additionally, the decoder may obtain a prediction block for a current block based on the at least one BV, or based on the at least one BV and a linear filter of a local illumination compensation (LIC) mode.

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 invention discloses a method and system for improving low signal-to-noise ratio direct sequence spread spectrum signal code tracking performance, and the method comprises the steps: generating an I-path bipolar code information source, and obtaining a baseband signal; performing convolution operation with the matched filter coefficient to obtain a matched filtering result; time lag and time lead are respectively carried out according to a specified length, so that a lag branch signal and a lead branch signal are respectively obtained; coherent integration is carried out on the local pseudo code sequence at a PL end, and an envelope value is solved; setting an incoherent accumulation length at a PS end, performing incoherent accumulation operation on the envelope value, and selecting a code phase discrimination mode to obtain a code phase discrimination value; and a phase change value is obtained through the linear filter and a loop filter of the code loop, and the phase change value is used for correcting the NCO of the local pseudo code sequence to realize closed-loop feedback. Code tracking performance is improved, signals can be accurately transmitted in a complex interference environment, error codes are reduced, inter-symbol crosstalk is effectively reduced, signal decoding accuracy is improved, and communication quality is guaranteed.

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.

The present invention discloses a high-precision spectral confocal microscopy imaging system, comprising a light modulation module, a spectrum splitting module, a scanning imaging module, an image detection device and a computer. The light beam emitted by the light modulation module passes through the scanning imaging module and is incident on a sample to excite a fluorescence signal. The fluorescence signal passes through the spectrum splitting module and is received by the image detection device. The image detection device converts the current signal into a digital signal and transmits it to the computer to complete imaging. The system is characterized in that the spectrum splitting module consists of a first converging lens, a first pinhole, a first linear long-wave pass filter, a second converging lens, a first linear short-wave pass filter and a third converging lens arranged in sequence. The system has the advantage of realizing high-precision spectral confocal imaging by combining lenses and linear filters on the basis of the traditional confocal microscopy imaging principle, solving the problem of spectral crosstalk in confocal imaging, and achieving a spectral resolution of less than 3.0 nm.

The present invention relates to a method for segmenting fundus image blood vessels based on linear filtering and deep learning, belonging to the field of medical image processing. The method includes: S1: Input the fundus image, and use the linear filtering algorithm based on the Hessian matrix to enhance the blood vessel area; S2: Use MobileNetV3 as the basic model of the blood vessel segmentation model, establish the segmentation network VSegNet, and then perform downsampling by adding an encoder based on the recursive module to the segmentation network VSegNet; S3: Add a decoder to the segmentation network VSegNet to perform upsampling and aggregation on the feature map output by the encoder; S4: When training the segmentation network VsegNet, calculate the loss value of the segmentation result by using the L1 norm of the segmentation prediction result and the segmentation ground truth image. The present invention strengthens the feature information extraction ability, thereby improving the model segmentation performance.

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 present invention discloses an adaptive sliding mode control method for a sway robot to resist false injection attacks, belonging to the technical field of sway robot control. The method comprises obtaining the initial trajectory tracking error equation of the sway robot based on the robot's dynamic model, a tracking trajectory reference signal, and the trajectory the sway robot needs to track; simulating the actuator attack signal using a neural network, obtaining the weight estimate of the neural network through a linear filter based on the Lyapunov stability theorem, and determining the actuator attack signal; determining the trajectory tracking error equation of the sway robot after being attacked by a false injection actuator based on the attack signal and the initial trajectory tracking error equation; and setting an adaptive integral sliding mode controller with the trajectory tracking error converging to 0 within a fixed time as the control goal to control the sway robot to achieve trajectory tracking. This method can ensure that trajectory tracking is not affected by attacks and interference, resolving the problem of insufficient protection against targeted attacks.

The present invention relates to the field of underwater vehicle inertial navigation technology, and specifically discloses an INS / DVL tightly coupled integrated navigation method, device, and system for underwater vehicles, comprising: constructing an INS / DVL tightly coupled integrated navigation model for the underwater robot; constructing a Mahalanobis distance based on the error covariance of the integrated model; determining a navigation state filter for the integrated model based on a comparison result of the Mahalanobis distance with a preset complex environment error determination threshold; executing a robust state filter based on a local consistency metric if the environment is complex; executing a basic nonlinear filter if the environment is non-complex; and determining the navigation parameters of the underwater robot based on the output of the filtered state. The INS / DVL tightly coupled integrated navigation method for underwater vehicles provided by the present invention can solve the problem of mismatch between noise parameters and actual conditions caused by environmental changes while avoiding parameter overcompensation in low-complexity environments and reducing computing power waste.

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.

This application relates to an audio signal enhancement method, performed by a computer device. The method including decoding received speech packets sequentially to obtain a residual signal, long term filtering parameters and linear filtering parameters; filtering the residual signal to obtain an audio signal; extracting feature parameters from the audio signal, when the audio signal is a feedforward error correction frame signal; converting the audio signal into a filter speech excitation signal based on the linear filtering parameters; performing speech enhancement on the filter speech excitation signal according to the feature parameters, the long term filtering parameters and the linear filtering parameters to obtain an enhanced speech excitation signal; and performing speech synthesis to obtain an enhanced speech signal based on the enhanced speech excitation signal and the linear filtering parameters.

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.