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127 results about "Least mean square algorithm" patented technology

The least mean square (LMS) algorithm is a type of filter used in machine learning that uses stochastic gradient descent in sophisticated ways – professionals describe it as an adaptive filter that helps to deal with signal processing in various ways.

Online decoupling identification method of multiple parameters of PMSM (permanent magnet synchronous motor)

The invention relates to the technical field of PMSMs, solves a coupling problem during online identification of multiple parameters of a surface-mounted type PMSM, and achieves online decoupling identification of PMSM inductance, stator resistance and rotor flux linkage. Accordingly, the technical scheme adopted by the invention is that an online decoupling identification method of multiple parameters of the PMSM comprises the steps as follows: 1) identifying and coupling analysis of parameters of the PMSM; 2) a decoupling identification strategy, wherein voltage deviation before and after D shaft current injection is used for increasing the order of a motor mathematical equation, so that the decoupling identification of multiple parameters of the surface-mounted type PMSM inductance, the stator resistance and the rotor flux linkage are achieved; 3) neural network identifier design, wherein according to a parameter online identification problem of the PMSM, online identification is performed on motor parameters by adopting a self-adaptive neural network structure and a weight convergence algorithm based on a least mean square algorithm. The method is mainly applied to the design and manufacture of the PMSM.
Owner:TIANJIN UNIV

Position observation device and method for rotor of built-in permanent magnetic synchronous motor based on adaptive filtering

The invention provides a position observation device and a position observation method for a rotor of a built-in permanent magnetic synchronous motor based on adaptive filtering, and belongs to the field of motor control, solving the problem of the convectional model method that six-time harmonic pulsation observation error is contained in the acquired rotor position angle observation value. The position observation device and the position observation method are respectively an adaptive notch filtering filter signal processing device and an adaptive notch filtering filter signal processing method based on the least mean square algorithm, and the device and the method are used for removing the six-time rotor pulsation error in the control technology model method applicable to a middle and high speed position-free sensor permanent magnetic synchronous motor. The method comprises the following steps of: acquiring equivalent back electromotive force information through the model method; regulating through the adaptive notch filtering filter based on the least mean square algorithm; then carrying out normalization processing on the back electromotive force information; and finally acquiring a rotor position observation value through a phase-locked loop. The device and the method provided by the invention are simple and easy to implement, can be used for effectively inhibiting the influence of the six-time pulsation error in the rotor position observation value and improving the performance of the position-free sensor permanent magnetic synchronous motor, and are applicable to a permanent magnetic synchronous motor control system.
Owner:哈工科讯(沈阳)智能工业技术有限公司

Method for improving performance of feedforward narrow-band active noise control system

InactiveCN101976560AImprove performanceReduce residual noise energy impactSpeech analysisSound producing devicesNoise controlSteady state
The invention discloses a method for improving the performance of a feedforward narrow-band active noise control system, which relates to the field of active noise control, and aims at the situation that the reference signal frequency acquired by a non-acoustic sensor in the feedforward narrow-band active noise control system with a minor access road on-line identification function and the target noise actual frequency are imbalanced and the problem the auxiliary noise brought by the minor access road on-line identification seriously hinders the reduction of the residual noise energy of the system. The control system comprises a frequency compensation subsystem, a signal synthesis subsystem and a minor access road on-line identification subsystem; the frequency compensation subsystem comprises a second-order autoregression module and a least mean square (LMS) algorithm module; the frequency compensation subsystem transmits the cosine component and sinusoidal component of the adjusted reference signal of an ith frequency channel to the signal synthesis; when the frequency imbalance reaches above 50%, the target noise can still be effectively restrained; and when in a steady state, the residual noise energy of the system is reduced to an ideal level, so that the performance of the control system can be further improved and is closer to the practice.
Owner:HARBIN INST OF TECH

Digital adaptive equalizer for T1/E1 long haul transceiver

The present invention relates to the implementation of a digital adaptive equalizer for a T1/E1 long haul transceiver which is capable of adapting to a wide range of cable types, cable lengths, and/or other data transmission impairments, particularly when the transmission path type and/or length are unknown. The digital adaptive equalizer contains two filter blocks, i.e., an IIR filter and a FIR filter, together with a filter selector block to select a best IIR filter based on an error estimation of the received data. Only a few sets of coefficients are found to be necessary to allow proper digital equalization of a large number of cable types and/or lengths. A filter selector block selects a desired set of coefficients corresponding to the optimum IIR filter. The coefficients may be programmed into volatile memory (e.g., RAM) or non-volatile memory (e.g., Flash). Alternatively, the coefficients may be hardwired into the IIR filter. The back end of the digital adaptive equalizer contains an adaptive finite impulse response (FIR) filter. In the disclosed embodiment, the FIR filter uses a least mean square (LMS) algorithm for adaptation to the unknown or changed T1 or E1 transmission channel or medium. The adaptive FIR filter adjusts the output from the IIR filter to accurately match the inverse response of the unknown channel used to transmit the received T1/E1 signal. Equalization may be temporarily frozen if periodic patterns are detected in the received T1/E1 signal. A restored T1 or E1 signal is output from the FIR filter, and thus from the digital adaptive equalizer.
Owner:INTEL CORP

Method for identifying signal and interference signal in ultrahigh frequency partial discharge detection in electrical equipment

ActiveCN102841294AAchieve and improve identificationImprove reliabilityTesting dielectric strengthMean squareTime delays
The invention discloses a method for identifying signals and the interference signals in ultrahigh frequency partial discharge detection in electrical equipment. The method comprises the following steps of: a. acquiring ultrahigh frequency signals near a device to be detected by using four sensors so as to obtain four paths of ultrahigh frequency partial discharge signals; b. carrying out the calculation of time delay by adopting a self-adaptive time delay calculation method based on a minimum mean square algorithm according to the obtained four paths of ultrahigh frequency signals, so as to obtain three time delay values; c. substituting the three time delay values obtained from the calculation into a positioning equation set, and solving the position of a discharge power supply by using a simulated annealing algorithm; and d. judging whether a detected signal comes from the interior or the exterior of the device according to a positioning result of the discharge power supply, so as to identify a partial discharge signal and an interference signal. By adoption of the method, the identification of the discharge signals and the interference signals in the partial discharge detection of the electrical equipment is realized and improved, and particularly the identifications to the discharge signals of periphery equipment and the interference of the equipment to be detected are realized.
Owner:STATE GRID HUBEI ELECTRIC POWER RES INST +2

Cable location system using magnetic induction

A method of electronically isolating a conductor by sensing a current signal in the conductor, adjusting the current signal to yield an adjusted signal, and applying the adjusted signal to the conductor to oppose the current signal. Amplifying the signal, as well as phase-shifting the signal may adjust the current signal. The current signal may be sensed using magnetic induction, and the adjusted signal applied to the conductor using magnetic induction as well. The method is carried out using a clamping device that can further evaluate an environment of the conductor, by randomly changing a magnitude and phase of a test signal applied to the conductor, and examining a received signal in response to the changes. A tracking error in the adjusted signal can be handled by predicting a next required phase shift. Prediction coefficients can be calculated utilizing a least means square algorithm. The invention is particularly applicable to the location of obscured (e.g., buried) conductors, and can be incorporated into a cable locator system that applies a trace signal to the target cable to be located. When this trace signal bleeds onto adjacent cables, the clamping devices of the present invention can be used to electronically isolate those cables, i.e., reduce or eliminate unwanted currents arising from coupling with the trace signal. A conventional receiver can then be used to trace the path of the target cable without interference from the nearby cables.
Owner:3M INNOVATIVE PROPERTIES CO

Multi -MIC noise reduction method used for mobile phone

The invention discloses a multi-MIC noise reduction method used for a mobile phone. The multi-MIC noise reduction method comprises the steps of S1, determining an MIC, which is closest to a terminal user, from all the MIC of the mobile terminal; S2, setting the MIC closest to the terminal user as a primary MIC and other MIC as secondary MIC; and S3, processing voice signals obtained by the primary MIC and the secondary MIC by using a noise reduction algorithm so as to obtain enhanced voice signals, wherein the step S3 specifically comprises the sub-steps of S301, performing voice endpoint detection for the voice signals obtained by the secondary MIC so as to obtain a pure noise segment and a noise-carrying voice segment, S302, self-adapting to the noise signals in the noise-carrying voice segment by adopting an improved LMS self-adapting filter and regarding the pure noise segment as the input in a least mean square algorithm of noise so as to obtain the voice signals being subjected to the self-adaptive filtration processing, and S303, performing spectral subtraction for the voice signals of the primary MIC and the voice signals being subjected to the self-adaptive filtration processing so as to obtain enhanced voice signals. The multi-MIC noise reduction method used for the mobile phone is good in noise reduction effect and can improve the tone quality of a mobile terminal.
Owner:HARBIN UNIV OF SCI & TECH

Fast self-optimization self-adaptive signal processing method and device of coherent communication technology

The invention discloses a fast self-optimizing self-adaptive signal processing method and processing device of a coherent communication technology as well as a coherent communication machine which comprises the processing device. The method comprises the steps: a signal is received; phase compensation is carried out on the signal; the signal is simplified and outputted through a diversity combiner; the signal that is simplified and outputted is processed through an equalizer; the signal outputted by the equalizer is carried out symbol judgment, the signal is outputted and simultaneously feeding back to the equalizer to generate an error signal. The device comprises a phase compensation unit, a diversity combiner unit, a self-adaptive judgment feedback equalizer unit, an error signal unit and a self-adaptive updating control unit that are sequentially connected in order. The self-adaptive updating control unit adopts a fast self-optimizing LMS algorithm to update the phase, the diversity combiner parameters and the equalizer parameters. The invention has a better signal processing performance, a lower bit error rate as well as a less calculation amount and saves the hardware cost.
Owner:INST OF ACOUSTICS CHINESE ACAD OF SCI

An aero-engine high-pressure rotor assembly error prediction method

The invention discloses an aero-engine high-pressure rotor assembly eccentricity prediction method. The method comprises the following steps: measuring radial and end jump error contour tracks of a matching surface; carrying out feature extraction on the contour track by using an image recognition method; establishing a complex domain autoregression model to obtain an error track feature vector; taking the seam allowance error feature vector as input, the eccentric position coordinate as output, the Gaussian function as a primary function of the intermediate layer, determining the center position of the hidden layer through K-means dynamic clustering, and estimating the weight of the output layer by using a minimum mean square algorithm to construct an RBF radial primary function neural network model; and substituting the measured data into the model for learning and training, performing error evaluation and model correction by using finite element simulation data, and determining neural network parameters so as to complete aero-engine high-pressure rotor assembly eccentricity prediction. According to the method, the seam allowance morphology error and the assembly deformation areconsidered, and the assembly eccentricity of the high-pressure rotor is predicted more quickly and more accurately.
Owner:XI AN JIAOTONG UNIV

Laser cloud-detection radar signal self-adaptive identification method based on least mean square algorithm

The invention relates to a laser cloud-detection radar signal self-adaptive identification method based on a least mean square algorithm, which synchronously compound two m sequences with different cycles so as to generate a compound pseudo-noise sequence and synchronously transmit the sequence to a continuous modulation laser transmitting terminal and a signal receiving terminal respectively; the signal receiving terminal carries out sectional accumulation average calculating operation to a received signal, then adopts an LMS algorithm to substitute the result of the sectional accumulation average calculating operation and the compound pseudo-noise sequence, and finally establishes a laser atmospheric scattering system model by utilizing the convergence result obtained by the LMS algorithm. The method effectively solves the circuit design problem caused by requirement on extremely high pulse peak value power and extremely narrow pulse when a pulse modulating laser radar is applied to the cloud-detection, and also the problems that the receiving terminal needs to collect a large quantity of samples for accumulation and average and the computation workload is large, the resolution is low, the signal-to-noise ratio is low and the like, thus greatly improving the measuring efficiency and accuracy.
Owner:ANHUI INST OF OPTICS & FINE MECHANICS - CHINESE ACAD OF SCI

Smart antenna self-adapting interference suppression method based on least square-lowest mean square

The invention relates to a smart antenna self-adapting interference suppression method based on least square-lowest mean square, which is an interference suppression algorithm based on training sequences; and by combining the least square algorithm with the lowest mean square algorithm, the method increases the velocity of convergence of the least mean square algorithm. The method includes the following steps: radio-frequency signals received by array antenna are converted into IF signals by a down converter; the IF signals are processed by the operations of analogue-to-digital conversion and digital down-conversion to obtain zero IF digital signals; the zero IF digital signals obtained in the step 2; and the local training sequence are utilized for carrying out corresponding processing to obtain local reference signals by computation delay; the low snapshot least square algorithm is utilized for calculating the initial weight vector of an aerial array; the weight vector calculated in the step 4 is used as the initial weight vector of the least mean square algorithm, and the least mean square algorithm is utilized for updating the aerial array weight vector; and the array weight vector calculated in the step 5 is adopted for the interference suppression of user data. The method achieves the purposes of increasing the availability of frequency spectrum of the system and reducing the complexity of the system.
Owner:CIVIL AVIATION UNIV OF CHINA

Method for suppressing acoustic feedback in a hearing device and corresponding hearing device

A hearing device (HD) incorporates a method for adaptive cancellation of acoustic feedback (AF). The method comprises generating an estimated feedback signal (EF) and subtracting the estimated feedback signal (EF) from the microphone signal (MS) before feeding it to a signal processor (SP) providing the primary hearing device function. The estimated feedback signal (EF) is generated in an adaptive filter (FE1), which is controlled using a least-mean-square algorithm, which operates on an error signal (E) and a reference signal (R). The algorithm may behave erroneously if the feedback path changes while a signal with low HF content, such as speech, is received. In this case, the hearing device (HD) will not be able to quickly adapt the HF characteristic of the adaptive filter (FE1) to the changed conditions. The adaptive filter (FE1) may thus have an incorrect HF gain when a subsequent signal with high HF content is received. This may lead to whistling or, alternatively, to an unwanted suppression of the subsequent signal. The problem is solved by modifying a filter function (H) applied to the error signal (E) and to the reference signal (R) in dependence on estimated relative amounts of high- and low-frequency signal content in the microphone signal (MS).
Owner:OTICON

Frequency-modulation continuous-wave radar self-interference elimination method based on fractional order least mean square

ActiveCN110146848AAutomatic adjustment of cancellation structureAdaptable to changeWave based measurement systemsICT adaptationSelf interferenceTarget signal
The invention discloses a frequency-modulation continuous-wave radar self-interference elimination method based on a fractional order least mean square. In the method, a frequency-modulation continuous-wave radar system uses a baseband de-sloping working mode, and operation of radio frequency domain self-interference cancellation is taken as a precondition. The method comprises the following stepsof firstly, carrying out multipath phase modulation on a baseband emission signal of a frequency-modulation continuous-wave radar to generate a reference signal; secondly, carrying out multipath amplitude modulation on the reference signal and generating a cancellation signal through multipath vector synthesis; then, before baseband de-sloping processing, carrying out self-interference cancellation on the cancellation signal and a radar receiving signal; and finally, using a fractional order least mean square algorithm to carry out feedback adjusting on multipath modulation coefficients so that the cancellation signal approaches the self-interference signal and a purpose of self-interference signal elimination is realized. In the invention, the self-interference signal in the radar receiving signal can be restrained and a target signal is not influenced so as to increase a detection probability and precision of the radar to a target.
Owner:XIDIAN UNIV

Time domain clutter suppression method suitable for LTE external radiation source radar

The invention discloses a time domain clutter suppression method suitable for LTE external radiation source radar. According to the method, an original monitoring signal is subjected to sinc functioninterpolation to obtain a monitoring signal linearly combined by limited integer delay reference signals; the interpolated monitoring signal and the reference signal are segmented separately; the segmented monitoring signal, the reference signal and a filter weight are subjected to fast Fourier transform, so that the signal is transformed from the time domain to the frequency domain; clutter suppression is performed according to the minimum mean square algorithm in the frequency domain, and adaptive variable step weight updating is carried out; and finally, the output results of each segment are combined. Compared with the prior art, the method has the advantages that accurate estimation of the fractional delay time is not needed, the interpolated monitoring signal and the reference signalare subjected to batch minimum mean square adaptive filtering; and when the estimated value is close to the real value, the output result is the required target echo signal, the clutter suppression performance of the monitoring channel is improved, and the target detection of a radar system is facilitated.
Owner:SHENZHEN RES INST OF WUHAN UNIVERISTY

Digital background correction method based on least mean square algorithm

ActiveCN110350918ASmallest highest bit capacitanceSolve the problem of invalid calibrationElectric signal transmission systemsAnalogue/digital conversion calibration/testingCapacitanceMean square
The invention discloses a digital background correction method based on a least mean square algorithm, which is suitable for a split SAR ADC. The method comprises the following steps: firstly, settinga split SAR ADC, and performing digital background correction based on a minimum mean square algorithm on the set split SAR ADC, wherein the split type SAR ADC comprises two ADC modules, a main DAC capacitor array in each ADC module adopts a non-binary capacitor array; meanwhile, the weight of the highest-order capacitor in the main DAC redundant capacitor array is set to be minimum, so that eachcapacitor including the capacitor with the maximum weight in the main DAC capacitor array can be effectively corrected through random switching of the correction DAC capacitor array, and the linearity and the dynamic range of the ADC are improved; the introduction of the correction DAC random switching mode can effectively solve the problem that the correction is invalid due to the fact that thetwo ADC capacitors are not consistent in direction. In addition, redundancy is introduced into the main DAC capacitor array, so that dynamic errors introduced into the quantization process of the system can be weakened, the correctness of each switching is ensured, and the iteration speed is increased.
Owner:UNIV OF ELECTRONICS SCI & TECH OF CHINA +1
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