34 results about "Amplitude response" patented technology

The invention provides a TMD lockout control method for motion suppression of a semi-submersible platform, and the method comprises the steps: carrying out the frequency domain decomposition of a random wave predicted by a target sea condition, obtaining the energy distribution of each frequency component, obtaining the equivalent amplitude of each frequency component according to the energy distribution, considering the amplitude response of the platform under the action of waves of different frequencies, and achieving the motion suppression of the semi-submersible platform. The method comprises the following steps of: calculating a weight ratio of platform motion response caused by each frequency component under the action of random waves by integrating three types of information, namely amplitude, amplitude response and motion suppression effect, and finally carrying out weighted combination on the locking time corresponding to each frequency component according to the weight ratio, so as to obtain the motion suppression effect of the platform. And the optimal locking control time suitable for the current random wave is obtained. According to the method, a high-complexity non-causal control command does not need to be deduced, control parameters can be updated in real time according to short-term sea condition changes, pitching response of a platform is effectively reduced, and comprehensive optimization control over multi-frequency waves is achieved.

Methods and systems for generating a sound. The method includes obtaining meshes representing at least one object in a frame of an environment, the environment including a source and a receiver; determining spatial continuity information and reflection normal information of the meshes; determining, based on the spatial continuity information and the reflection normal information, reflection paths between the source and the receiver involving the at least one object, each of the reflection paths having at least one reflection point associated with the at least one object; obtaining spatially sampled results by spatially sampling a space around the at least one reflection point of reflection paths using multiple distributions of rays, the spatially sampled results correlating with geometric information of the meshes; generating reflection amplitude responses for each of multiple audible frequencies in the environment based on the spatially sampled results; and producing a sound based on the reflection amplitude responses.

The invention discloses a millimeter wave dual-polarization phased-array antenna beam forming method and device, and relates to the technical field of high-frequency communication, and the method comprises the following steps: obtaining a complex weight signal of each array element in an antenna array under a horizontal polarization channel and a vertical polarization channel, respectively carrying out amplitude response and phase response acquisition and calibration processing on the complex weight signal; performing dynamic extraction and modeling on amplitude errors and phase errors of each array element in different polarization channels; on the basis of error estimation, constructing a machine learning prediction module comprising a time sequence prediction model; performing joint estimation on a real-time error estimation result and a prediction error result; constructing a hardware-software collaborative calibration structure comprising a dynamic gain control module on the basis of the compensation weight; beam focusing of the target space area is realized; according to the invention, the hidden problems of beam offset, sidelobe lifting, gain loss and the like caused by inconsistency of polarization channels are solved.

Various current measurement systems and methods of signal processing are disclosed. In one example, there is a current measurement system that includes a filter with a corner frequency within the operating frequency range of the current measurement system. This provides a system with good SNR, whilst also preventing or reducing the likelihood of output saturation during large di / dt spikes. The system further includes an equaliser arranged to compensate for the phase and / or magnitude response of the filter within the operating frequency range.

The invention discloses a shallow lake turbidity remote sensing inversion method and system based on wind drive physical constraint, and the method comprises the steps: constructing a time sequence pairing sample through combining a remote sensing image and actually measured data, and extracting wind drive characteristics such as wind speed, wind stress and strong wind duration; establishing direction consistency priori by using monotone calibration to generate an expected direction signal, and constructing a wind-driven amplitude response surface interpolation to obtain an expected change amplitude; in model training, a double-moment regression network sharing weights is adopted, and an actual measurement supervision error and wind-driven physical constraints are jointly optimized; and finally, carrying out pixel-level reasoning and verification on the multi-temporal remote sensing image to realize continuous estimation of spatial distribution and dynamic change of turbidity. According to the method, a double-moment input and shared weight network structure is introduced, the physical mechanism and deep learning advantages are fused, the time sequence continuity, the physical interpretability and the cross-space-time generalization ability are remarkably improved while high inversion precision is kept, and an efficient and reliable technical approach is provided for water environment remote sensing monitoring.

An antenna tuner receives outputs of detectors coupled to an input and / or an output of each power amplifier (PA) of a plurality of PAs of a multiple antenna system. A processor of the antenna tuner is configured to process the detector outputs and to generate at least one of a gm gate bias value and a cascode gate bias value to be applied to gm and cascode gates, respectively, of a last stage of each of the PAs to change the shape of an input amplitude-to-output amplitude (AM-AM) response of each of the PAs independently of one another. The processor can also be configured to process the detector outputs to generate a varactor bias value to be applied to a terminal of a varactor of each of the PAs to change the shape of an input amplitude-to-output phase shift (AM-PM) response of each of the PAs independently of one another.

The invention relates to the technical field of power equipment state evaluation, and discloses a three-phase transformer state evaluation method, system, equipment and medium, and the method comprises the steps: collecting the frequency-amplitude response data of each phase winding of a three-phase transformer; dividing at least two characteristic sensitive frequency bands, wherein the frequency bands respectively correspond to core components of the transformer; extracting difference characterization characteristic parameters of each phase and each frequency band, and calculating a three-phase imbalance coefficient; and generating a dynamic threshold value based on the operation condition of the transformer and historical data, performing multi-level state evaluation in combination with the deviation degree of the unbalance coefficient and the threshold value, and positioning an abnormal phase. The method breaks through the limitation of a traditional fixed threshold value and a single feature, adapts to different operation scenes through the dynamic threshold value, expands feature parameter types and frequency band configuration flexibility, achieves state accurate grading and anomaly tracing, solves the problems that a traditional method is poor in adaptability and narrow in protection range, and remarkably improves the evaluation universality and the intelligent level.

A dynamically controllable metasurface antenna system is disclosed. The system comprises: a metasurface comprising a plurality of electronically addressable meta-atoms, each meta-atom comprising an integrated electronic component configured to control a local phase and / or amplitude response in response to an applied bias; a feed configured to launch surface waves onto the metasurface; and a control circuit operatively coupled to the plurality of meta-atoms and configured to selectively bias individual meta-atoms to dynamically control at least one of frequency response, beam direction, or polarization state of the antenna system in real time.

The embodiment of the application provides a kind of automation test control method and system of multi-channel amplitude-phase amplification module, belong to electronic test technical field.The method comprises: obtaining the amplitude response data and the phase response data of each channel in multi-channel amplitude-phase amplification module under preset test excitation condition, and constructs channel reference parameter set;Sequence influence model is constructed, and the test execution order of each channel is optimized;The amplitude-phase test is executed to each channel, the amplitude response data and the phase response data of each channel at current time are obtained, and virtual reference channel parameter is constructed;The amplitude response data and the phase response data of each channel are corrected in consistency, and the corrected amplitude-phase parameter of each channel is obtained, and the amplitude-phase test result corresponding to each channel is generated based on corrected amplitude-phase parameter.The application scheme realizes the unified alignment and error suppression of multi-channel amplitude-phase test result, improves test consistency and stability.

A correction device (100) for a multi-channel time-interleaved ADC system (300) is disclosed. The correction device (100) is configured to correct amplitude response mismatch errors of an ADC output sequence v using a first time-varying function B and further to correct phase response mismatch errors of the ADC output sequence using a second time-varying function C. Thus, due to the use of the first time-varying function B and the second time-varying function C, low complexity converter performance is improved.

The application relates to the technical field of short-wave communication and provides a 3G-ALE burst waveform identification method, device and electronic equipment based on a correlation peak fusion graph. The method comprises the following steps: S1, according to the protocol frame structure of various burst waveforms in a 3G-ALE physical layer, extracting preset fields irrelevant to service payloads in the various burst waveforms, and constructing complex baseband template sequences corresponding to the various burst waveforms; S2, obtaining a complex baseband observation sequence corresponding to a 3G-ALE burst signal to be identified; S3, performing cross-correlation calculation on the complex baseband observation sequence and the complex baseband template sequences corresponding to the various burst waveforms respectively, and obtaining correlation amplitude responses corresponding to the various burst waveforms; S4, mapping all the correlation amplitude responses to corresponding predetermined regions in a graph according to a preset spatial layout, and obtaining a correlation peak fusion graph; and S5, inputting the correlation peak fusion graph into a trained 3G-ALE burst waveform identification model, and obtaining the waveform category of the 3G-ALE burst signal.

The invention discloses a sound compensation method and device, an earphone and a storage medium, and the method comprises the steps: responding to the condition that the earphone is worn on a human ear, playing a test audio signal through a moving-iron loudspeaker and a moving-coil loudspeaker, and collecting an ear canal signal of the test audio signal through a feedback microphone; determining an amplitude response curve of a first frequency band based on the test audio signal and the ear canal signal, and determining a response error curve based on the amplitude response curve; determining time delay information of the ear canal signal based on a plurality of candidate crossover frequencies; and determining a plurality of loss values based on the plurality of candidate crossover frequencies, the response error curve and the time delay information, determining a digital filtering coefficient based on the plurality of loss values, and performing sound compensation based on the digital filtering coefficient. And the accuracy of sound restoration and the consistency of hearing feeling are obviously improved.

The application discloses a signal receiving method for spread spectrum underwater acoustic communication of long-distance time-varying channel, which comprises the following steps: obtaining the received communication signal, carrying out down-conversion and low-pass filtering to obtain a baseband signal, and carrying out synchronization to divide the baseband signal into pilot signals and spread spectrum sequence signals; using the pilot signals to complete initial channel estimation to obtain initial Doppler estimation value, time delay and amplitude response estimation value of each path; using the estimated Doppler to carry out frequency offset compensation and resampling on the first spread spectrum sequence receiving signal; using the estimated time delay and amplitude response to construct a first to-be-estimated spread spectrum sequence signal Rake receiver; combining the Rake receiver and using maximum likelihood estimation to realize the decision of the current spread spectrum sequence and the log-likelihood ratio of each bit; using the decided spread spectrum sequence to estimate the Doppler, time delay and amplitude response of the current spread spectrum sequence signal; and in this way, the decision of all spread spectrum sequences is completed. The application can effectively reduce the demodulation error rate.

The present application relates to a kind of LFMs based on amplitude-phase response measurement timing method, belong to telemetry, radar and test measurement technical field, in the measurement method of amplitude-phase response, it is usually based on the way of single tone or double tone scanning and time-consuming is longer, and single tone cannot measure system phase response;And based on the way of multi-carrier, due to power limit, leading to the signal-to-noise ratio of each sub-carrier reduces, measurement result is poor. Again due to the related advantages of LFM signal, such as bandwidth, energy concentration and pulse compression etc., it gradually obtains application in this kind of measurement calibration system. After using the timing estimation of the present application, not only the amplitude response can be accurately measured, but also the phase response can be accurately measured. The actual measurement results show that the phase measurement curve of each time is very close to even coincide, so the idea of average noise reduction can be further used to improve the measurement accuracy of phase response.

The present application is suitable for the field of geophysical exploration technology, and provides a rock physics driven time-frequency domain P-wave velocity dispersion attribute prediction method, comprising the following steps: establishing a rock physics model; taking the rock physics model as the driving force, inverting the micro-fracture porosity in the well, calculating the skeleton elastic modulus of the rock containing pores and fractures, and predicting the frequency-dependent P-wave velocity logging curve; constructing a time-frequency domain P-wave velocity seismic inversion initial model; constructing a TF-AVO theoretical formula, developing a time-frequency domain P-wave velocity seismic inversion method based on the L-BFGS algorithm, and taking the initial model as the prior information to perform P-wave velocity inversion; analyzing the relative change of the frequency-dependent P-wave velocity under different frequencies, quantitatively predicting the P-wave velocity dispersion attribute; based on the logging data, calibrating the P-wave velocity dispersion attribute prediction result, and detecting the reservoir oil and gas. The present application has higher precision and resolution, significantly improves the stability and reliability of the P-wave velocity dispersion attribute prediction, and overcomes the dependence on strong amplitude response.

The invention discloses a variable fractional delay filter optimization method and system based on full coefficient compression. The method comprises the following steps: firstly, constructing a variable fractional delay filter frequency response function based on a Farrow structure; then setting an amplitude constraint based on a variable amplitude response peak error and a delay constraint based on a variable fractional delay peak error; setting constraint parameters based on a full-coefficient compression L0 norm, constructing an optimization model based on a full-coefficient compression sensing L0 norm, and converting the optimization model into an L1 norm optimization model; and finally, solving a weighted L1 norm optimization problem to obtain an optimal variable fractional delay filter coefficient. The system is used for realizing the variable fractional delay filter optimization method based on full coefficient compression. According to the method, the variable fractional delay filter with the optimal amplitude error performance can be obtained under the condition of least resource and time consumption, the amplitude error performance of the system is optimized, and the method is suitable for the field of radar signal processing.

The present disclosure relates to a distributed optical fiber sensing data amplitude response recovery method and device, and a computer device, a computer readable storage medium and a computer program product, the distributed optical fiber sensing data amplitude response recovery method comprising: obtaining DAS data obtained by a sampling channel sampling seismic source and an observation system parameter for the DAS data; calculating an incident angle θ corresponding to each seismic source for each sampling channel according to the observation system parameter; calculating an amplitude recovery coefficient corresponding to the sampling channel sampling seismic source according to each incident angle θ respectively; for each sampling channel sampling seismic source, performing amplitude recovery processing on the DAS record corresponding to the sampling channel sampling seismic source by using the corresponding amplitude recovery coefficient, to obtain recovered DAS data; and outputting the recovered DAS data. The amplitude response recovery method of the DAS data proposed by the present disclosure realizes effective recovery of the amplitude response of the DAS data.

Embodiments of the present disclosure provide a method, device and equipment for evaluating inertia and primary frequency modulation capability of a power supply node. The method comprises injecting a small disturbance power into a power system and calculating impedance at each power supply node in the power system; drawing an amplitude response curve corresponding to a frequency response transfer function according to the impedance at the power supply node and a frequency response transfer function formula at the power supply node; and calculating inertia coefficients and primary frequency modulation capability coefficients of each power supply node according to each amplitude response curve. In this way, the inertia coefficients and the primary frequency modulation capability coefficients of each power supply node can be calculated in real time from the perspective of impedance under the premise of ensuring normal operation of the power system, and the inertia and the primary frequency modulation capability of the power supply node can be quickly and accurately evaluated.

The invention provides a positive and negative sequence electric quantity separation method and device. The method comprises the steps of obtaining an instantaneous electric quantity signal in a three-phase power grid; performing phase angle deviation of a preset angle on the instantaneous electric quantity signal based on an all-pass filter to obtain an instantaneous orthogonal electric quantity signal; and determining a positive sequence component and a negative sequence component of the instantaneous electric quantity signal according to the instantaneous electric quantity signal and the instantaneous orthogonal electric quantity signal. The phase angle deviation of the preset angle is performed on the instantaneous electric quantity signal based on the all-pass filter so as to acquire the instantaneous orthogonal electric quantity signal, and the phase angle deviation can be realized with extremely high precision by utilizing the characteristic that the all-pass filter keeps the amplitude response constant in the whole frequency range and only changes the phase of the signal, so that the error caused by the amplitude change is avoided. According to the method, the instantaneous orthogonal electric quantity signal is obtained firstly, then the positive sequence component and the negative sequence component are determined in combination with the instantaneous electric quantity signal, the positive sequence component and the negative sequence component are rapidly obtained in a three-phase power grid system with the high real-time performance requirement by means of the orthogonal relation between the signals, and error analysis is facilitated.

The application provides a kind of laser light source line width stability multi-parameter joint evaluation method and system, obtains the junction temperature parameter of laser, environmental sound spectrum data and optical interference original signal during power mutation;The energy accumulation operation is carried out to the environmental sound spectrum data, and acoustic characteristic energy value sequence is generated;The optical interference original signal is phase demodulation processing, and line width drift value sequence is generated;The junction temperature parameter, the acoustic characteristic energy value sequence and the line width drift value sequence are carried out frequency domain correlation operation, and joint transfer function matrix is generated;The amplitude response modulus of the matrix is calculated, and dynamic weight vector is obtained;The line width drift value sequence is Fourier transformed, and initial line width frequency domain distribution vector is obtained, combined with the dynamic weight vector, and multi-parameter joint evaluation value is calculated;The application realizes dynamic weighted comprehensive evaluation to laser light source line width stability, and improves the accuracy and robustness of evaluation.

This invention discloses a signal receiving method for spread spectrum underwater acoustic communication in long-distance time-varying channels, comprising: acquiring the received communication signal, obtaining a baseband signal through down-conversion and low-pass filtering, and dividing the baseband signal into blocks according to pilot signals and spread spectrum sequence signals after synchronization; performing initial channel estimation using the pilot signals to obtain initial Doppler estimates, time delay estimates, and amplitude response estimates for each path; performing frequency offset compensation and resampling on the first spread spectrum sequence received signal using the estimated Doppler; constructing a Rake receiver for the first spread spectrum sequence signal to be estimated using the estimated time delay and amplitude response; combining the Rake receiver and using maximum likelihood estimation to realize the decision on the current spread spectrum sequence and the log-likelihood ratio of each corresponding bit; estimating the Doppler, time delay, and amplitude response of the current spread spectrum sequence signal using the decided spread spectrum sequence; and so on, completing the decision on all spread spectrum sequences. This invention can effectively reduce the demodulation bit error rate.

The utility model discloses a portable experiment demonstration teaching device for measuring the inherent frequency of an object, and the device comprises a pedestal, a sound box which is disposed on the pedestal and is used for outputting a sound wave signal, and a sound collection frame. The beneficial effects are that a cooperative reflection structure of the sound collection rack and the sound gathering rack is additionally arranged so that directional reflection and energy focusing can be carried out on multi-frequency sound waves emitted by the sound box, sound wave energy is enabled to act on the surface of a measured object more densely, the amplitude response speed of the measured object in a resonance state is enabled to be increased by improving sound energy transmission efficiency, and the sound quality of the measured object is improved. And meanwhile, environmental noise interference is effectively inhibited, the signal-to-noise ratio of signals is improved, the time consumed for capturing the resonance peak value by the oscilloscope is shortened, and rapid locking of the inherent frequency is realized in cooperation with the real-time frequency regulation and control function of the smart phone APP, so that the experiment time is effectively shortened, the experiment precision is improved, and demonstration teaching is facilitated.

The invention relates to a signal filtering method, device and equipment, a medium and a program product in the technical field of signal processing, and the method comprises the steps: determining a target transfer function of each parameter equalizer in a graphic equalizer according to a sampling rate, each center frequency and an analog amplitude at each center frequency; determining a target gain of each parameter equalizer according to the target transfer function and the expected amplitude response at each center frequency; determining a target filter coefficient of each parameter equalizer according to the target transfer function, the center frequency, the sampling rate, the preset bandwidth and the target gain of each parameter equalizer; taking the target filter coefficient corresponding to each parameter equalizer as a target, and adjusting the function expression of each parameter equalizer to obtain a target graphic equalizer; and inputting the obtained signal to be filtered into the target graphic equalizer to obtain a target signal output after filtering of the target graphic equalizer. And the accuracy and convenience of signal filtering are improved.

The invention discloses a corrosion defect quantitative evaluation method fusing spatial and temporal distribution of multiple acoustic emission sources, and belongs to the field of nondestructive testing. The method comprises the following steps: carrying out acoustic emission event judgment on collected multichannel acoustic emission original signals, and screening effective acoustic emission event signal segments; preprocessing all the effective acoustic emission event signals, including band-pass filtering, window function processing and adjustable Q factor wavelet transform, and performing time reversal focusing analysis to obtain global amplitude response distribution of each event signal and obtain an acoustic emission source position corresponding to each event signal; all global amplitude responses are accumulated, the boundary contour and the geometric parameter set of the corrosion area are extracted through a spatial clustering algorithm and convex hull fitting after binarization processing, and the boundary contour and the parameter set of the corrosion area are obtained. According to the method, information of a plurality of acoustic emission events in a section of signal is effectively fused, the accuracy of corrosion defect positioning is improved, and extraction and quantitative characterization of a corrosion defect contour are realized.

The invention discloses a self-interference elimination circuit. The self-interference elimination circuit comprises an input attenuation module, a variable gain delay array and an output attenuation module. The input attenuation module is used for regulating and controlling the power of a self-interference signal generated by a transmitter; the variable gain time delay array is used for performing high-precision time delay frequency characteristic and amplitude frequency characteristic regulation and control on the self-interference signal; the output attenuation module is used for regulating and controlling the output power of the variable gain delay array; the self-interference elimination circuit establishes a self-interference elimination signal by simulating group delay, phase and amplitude response of a self-interference signal in a propagation process, and realizes self-interference elimination at a radio frequency front end. The self-interference elimination circuit can be used for self-interference elimination of the radio frequency front end and has the advantages of being large in delay range and high in adjustment precision.

The invention provides a stroke calibration method of a voice coil motor, which comprises the following steps of: respectively applying a minimum driving current, a large driving current and a zero driving current to the voice coil motor to obtain digital output quantities Hn, Hp and Hz of magnetic field intensity obtained by a position sensor under each driving current; a gain gain3 is obtained through calculation; calculating a target position hallcal corresponding to the digital output quantity hall output by the position sensor; periodic signals with different frequencies are applied to the voice coil motor, the average value A of the amplitude response of the voice coil motor under different frequencies is obtained, and then the gain value gainf of the amplitude response is obtained; and obtaining a lens position fed back to a control algorithm based on a target position hallcal corresponding to the digital output quantity hall output by the position sensor and the gain value gainf of the amplitude response. Therefore, the control precision of the lens is improved.

The invention relates to a hearing aid omnidirectional listening speech enhancement method and system based on a microphone array, and belongs to the technical field of hearing aids, and the method comprises the steps: respectively determining forward and backward wide beam signals according to the frequency domain representation of forward and backward microphones, and respectively constructing forward and backward amplitude response functions based on the forward and backward wide beam signals; and judging the time-frequency energy distribution of the microphone acquisition signal based on the ratio of the forward and backward amplitude response functions so as to obtain a forward narrow beam signal and a surrounding wide beam signal. And estimating environmental noise of the forward narrow beam signal and the surrounding wide beam signal through a minimum power spectrum signal in a set time window to determine whether target speech exists, and outputting a noise power spectrum. According to the method, the target voice signal power spectrum is estimated according to the noise power spectrum, and the Wiener filtering technology is called for noise elimination to obtain the synthesized target voice signal amplitude spectrum after noise reduction, so that the naturalness of an auditory scene and the flexibility of a sound source are considered while high voice gain is ensured.

This disclosure is directed to noise mitigation for processing circuitry of an electronic device having a neural network circuit. A neural network circuit may include multiple neural network layers (e.g., a transformer) to perform inference operations. The processing circuitry may include a noise mitigation circuit to reduce a magnitude or magnitude response of noise signals, for example, at one or more frequencies. The noise mitigation circuit may reduce an amplitude of the noise signals by aperiodically providing different clock signals having different clock frequencies to the neural network layers during an inference operation. Alternatively or additionally, the noise mitigation circuit may reduce an amplitude of the noise signals by aperiodically changing clock signal frequencies and supply voltage values of the neural network layers during the inference operation.

The application relates to a shipborne ground wave radar target direction finding method based on multi-time amplitude-frequency information of a quadrature magnetic antenna system, and belongs to the technical field of high-frequency ground wave radar direction finding. The method obtains radar time domain data, target detection results and attitude data of a shipborne platform during a change in a bow direction, and extracts target echoes according to the target detection results; time-frequency analysis is performed on the target echoes to obtain a Doppler and amplitude information time sequence, and the Doppler and amplitude information time sequence is framed according to a fixed frame length to construct a single-frame Doppler frequency shift change sequence. The amplitude response characteristics of two channels of the quadrature magnetic antenna are utilized, and a target azimuth initial estimation value is obtained by using a amplitude comparison algorithm; further, a theoretical Doppler frequency shift change sequence is generated by combining the attitude data, and single-frame azimuth accurate estimation is obtained by matching search through a least square criterion, and multi-frame results are smoothed to output a stable azimuth. The method has the advantages of high direction finding accuracy and strong robustness under the condition of target echo broadening.

The invention relates to the technical field of control or regulation systems, and discloses a control or regulation system for response characteristic calibration of logging equipment, which comprises a frequency spectrum characteristic extraction module, a topological form locking controller, a thermodynamic benchmark evolution module and an amplitude response calibration controller, a frequency spectrum feature extraction module is used for extracting power values of intrinsic background noise in a passband core frequency band and a stop band edge frequency band, a dimensionless frequency spectrum shape factor is constructed, a topological form locking controller adjusts bandwidth control parameters according to deviation between the factor and a topological constant reference, a physical bandwidth is locked, and the bandwidth is locked. On the premise of bandwidth locking, the amplitude response calibration controller adjusts gain in combination with the deviation between the theoretical target power value calculated by the thermodynamic reference evolution module and the actually measured total power, and the bandwidth is locked by using the intrinsic noise spectrum shape invariant through a double-closed-loop time sequence decoupling mechanism. And the problems of model mismatch and multivariable coupling control caused by no external standard source and aging of devices are solved.