114 results about "Gain factor" patented technology

The invention discloses a multi-channel AFE signal acquisition method and system based on adaptive adjustment, and the method comprises the steps: synchronously acquiring multi-channel analog signals through a multi-channel parallel AFE module, and configuring an independent programmable gain amplifier and an anti-aliasing filter for each channel; extracting time-frequency domain characteristic parameters of each channel signal in real time; dynamically generating an optimal adjustment strategy based on the time-frequency domain characteristic parameters, and adaptively adjusting the gain coefficient, the filtering bandwidth and the sampling rate parameter of each channel through a closed-loop feedback mechanism; and abnormal signal detection is realized by adopting an inter-channel cross validation algorithm, and automatic isolation of a fault channel and intelligent switching of a redundant channel are triggered. Through dynamic parameter intelligent adjustment, multi-channel cooperative fault tolerance and energy efficiency optimization architecture, the problems of signal distortion, response lag and high power consumption in a traditional multi-channel signal acquisition system are solved, and the signal fidelity and the system reliability are remarkably improved.

The invention discloses an active disturbance rejection controller control method and device, and the method comprises the steps: receiving an actual output feedback signal of a controlled object and a historical output control signal of a controller based on a linear expansion state observer, calculating the total disturbance of a system in real time through a three-order state-space equation, and obtaining a disturbance estimation value; receiving the disturbance estimation value and the control input gain coefficient based on a compensator, calculating a dynamic compensation amount for offsetting the total disturbance of the system, and outputting the dynamic compensation amount to a signal synthesis node; a deviation signal of a set value signal used for defining the target state of a controlled object and a feedback signal actually output by the controlled object is received based on a controller, and an anti-interference basic control signal is generated in combination with the bandwidth of the controller; and subtracting the dynamic compensation amount from the anti-interference basic control signal based on the signal synthesis node to generate an active-disturbance-rejection control amount, outputting the active-disturbance-rejection control amount to a controlled object, and suppressing external interference and internal parameter drift of the controlled object.

The invention provides an automatic accurate correction method and system for a sound equalizer, and relates to the technical field of sound regulation and control, and the method comprises the steps: obtaining a standard audio signal; obtaining a secondary test signal; obtaining a fitting target signal; obtaining a correction signal based on the standard audio signal, the secondary test signal and the fitting target signal; acquiring a center frequency sequence, a compensation center frequency sequence, a gain coefficient sequence, a compensation gain coefficient sequence and a bandwidth information sequence of the equalizer; calculating a filter coefficient and a frequency response curve of the filter; obtaining an actual optimal gain sequence, and calculating a correction filter coefficient; and inputting the correction filter coefficient to a second-order IIR filter to realize accurate correction of the sound field. The technical problems that in the prior art, interference of human factors is difficult to avoid, subjective deviation is likely to be generated, and efficiency is low are solved. The technical effects of high efficiency, high precision, no interference of human factors and capability of supporting personalized adjustment of the sound equalizer are achieved.

The invention provides a signal processing method and device which can be applied to the technical field of signal processing. The signal processing method comprises the steps of performing feature analysis on a to-be-processed signal, and determining a target signal category from a plurality of predetermined signal categories; determining a target gain coefficient from a plurality of predetermined gain coefficients according to the target signal category; adjusting the initial waveform of the to-be-processed signal by using the target gain coefficient to obtain a target adjustment signal; performing a circuit transmission simulation test on any predetermined signal in any predetermined signal category to obtain an initial output signal; compensating the frequency spectrum of any predetermined signal according to the initial gain function and the initial output signal, and performing a circuit transmission simulation test on the compensated signal to obtain a compensated output signal; and performing error verification on the compensation output signal based on any predetermined signal, and performing adjustment and conversion processing on the initial gain function according to a verification result to obtain a predetermined gain coefficient corresponding to any predetermined signal category.

The invention discloses a noise simulation method and device of a transformer, a storage medium and electronic equipment. The method comprises steps that geometric information of a reference emission surface of a target transformer, an operation condition of the target transformer, and distance information between the target transformer and a sound measurement point are acquired, and the reference emission surface is a main radiation area of sound waves of the target transformer; based on the geometric information and the distance information, the sound pressure level of the target transformer under the operation condition is determined, and the sound pressure level is used for representing the sound pressure level of the simulation noise at the sound measurement point; based on the sound pressure level and the operation condition, adopting a pre-established standard spectrogram library to generate initial simulation noise; and performing correction processing on the initial simulation noise by adopting the gain coefficient to obtain target simulation noise of the target transformer. The technical problem that the accuracy of the generated analog noise of the transformer is low in the prior art is solved.

The invention relates to the technical field of elevator security and protection, in particular to an elevator remote alarm talkback noise reduction method and system.The method comprises the steps that noisy voice signals are processed, 40-dimensional voice features are extracted, the 40-dimensional voice features comprise 20 Mel-frequency cepstral coefficients, and first derivatives and second derivatives of the first 10 Mel-frequency cepstral coefficients are taken in sequence; inputting the 40-dimensional speech features into a pre-trained RNN (Recurrent Neural Network) model, wherein the RNN model outputs 20 gain coefficients; and dynamically adjusting coefficients of 20 first-order IIR band-pass filters based on the 20 gain coefficients, and after the noisy voice signal passes through the 20 first-order IIR band-pass filters in parallel, adding outputs of the first-order IIR band-pass filters to obtain a noise-reduced voice signal. According to the method, the technical problems of poor effect, low efficiency or high cost of an existing elevator remote alarm talkback noise reduction method can be solved in a comprehensive noise reduction mode combining the neural network and traditional signal processing, and efficient noise reduction is achieved on low-cost hardware.

The invention relates to the technical field of online material performance test instruments, and discloses a sensor pulse pressure fatigue test closed-loop control system based on signal drift compensation, which comprises a load excitation execution module, a signal feedback acquisition module and a closed-loop control logic module, the closed-loop control logic module selects a phase-locked characteristic observation window by calculating a load first-order derivative, extracts a dynamic gain coefficient reflecting the response characteristic of a controlled test object in the observation window, and inputs the dynamic gain coefficient into a filter to generate a characteristic variable; the variable is compared with the standard response envelope line, the deviation is calculated, the compensation factor is calculated and superposed to the driving instruction, the real-time correction of the output gain is realized, and through the identification and active compensation of the signal drift, the sensing deviation caused by the structural degradation of the sensitive component is effectively eliminated, and the sensing consistency of the test process is maintained.

The invention belongs to the technical field of power systems, and provides a frequency estimation system and method based on a composite second-order generalized integrator-frequency-locked loop. The frequency estimation system based on the composite second-order generalized integrator-frequency-locked loop comprises an i-phase two-stage series second-order generalized integrator-orthogonal signal generator, the input signal of which is an i-phase voltage signal vi and an estimation frequency fed back by the frequency-locked loop, and is configured to convert vi into a signal vi'with the same frequency and the same phase as vi, simultaneously obtaining a signal qvi'and a voltage estimation error epsilon vi, wherein the frequency of the signal qvi 'is the same as that of the vi', and the phase lag is 90 degrees; the frequency-locked loop gain normalization module is configured to normalize the output signal; multiplying the voltage estimation error epsilon vi by the signal qvi'and the adaptive gain to obtain a frequency error variable of a corresponding phase; the phase-locked loop is configured to add the frequency error variables of the alpha phase and the beta phase, then multiply the frequency error variables of the alpha phase and the beta phase with the gain constant coefficient-gamma and the normalized gain coefficient A, and then perform integration to obtain the estimated frequency.

The invention provides a steer-by-wire angle signal loss adaptive gain compensation method, which comprises the following steps: synchronously acquiring real-time angle signal sets of a hand feeling simulator and a steering actuator, carrying out signal loss detection and fault grading on the real-time angle signal sets to obtain a fault grade, and generating a redundancy switching strategy based on the fault grade; acquiring a historical angle signal set of a plurality of past sampling periods according to a redundancy switching strategy, and calculating an angle compensation amount according to the historical angle signal set; acquiring real-time vehicle parameters, generating a time-varying gain coefficient based on the real-time vehicle parameters, and calculating an adaptive gain compensation angle by using the time-varying gain coefficient and the angle compensation amount; and acquiring historical motor current of a past sampling period, calculating motor output current according to the adaptive gain compensation angle, performing gradient limitation on the motor output current by using the historical motor current, and performing steering control according to the motor output current after gradient limitation.

The invention relates to the technical field of current calculation, and discloses a digital compensation grounding current calculation method and device, terminal equipment and a storage medium. The method comprises the following steps: when a current signal of any phase of a three-phase current is obtained, obtaining a sampling value, a gain coefficient and a bias coefficient of a channel where the current signal is located and a sampling moment of the current signal; firstly calculating the bias error of the bias coefficient, then calculating the gain error of the gain coefficient, and finally calculating the phase error at the sampling moment; and compensating the sampling value according to the bias error, the gain error and the phase error to obtain a compensated three-phase current, and calculating a grounding current based on the compensated three-phase current.

The application discloses a method and a control system for improving the weak power grid adaptability of a grid-connected inverter, and belongs to the field of grid-connected inverter control. The method relates to a control system composed of a main control loop of a novel current controller and a point of common coupling voltage feedforward branch. First, reference currents pass through a first proportional controller, a second proportional controller and a delay module respectively, the outputs of the first proportional controller and the delay module are added by a first adder to obtain a first output signal; meanwhile, a sampling point of common coupling voltage passes through a virtual capacitor link and a repetitive prediction link to generate a second output signal. The first output signal and the second output signal are added by an adder, and then divided by an inverter bridge gain coefficient to obtain a modulation wave. The modulation wave passes through a PWM module to generate a driving signal to drive power devices of the inverter, so as to control an in-grid current.

The invention describes a method for noise reduction in a hearing instrument (1), wherein an acousto-electrical input transducer (E1) of the hearing instrument (1) generates an input signal (E1) from ambient sound (2), wherein a frequency-bandwise noise reduction (21) is applied to a processing signal (V1) derived from the input signal (E1), in which stationary noise in the respective frequency band (Bj, Bk) is detected, and depending on the detected stationary noise, a gain factor (Gk) of the processing signal (V1) for the respective frequency band (Bj, Bk) is set, wherein an analysis (12) is applied to the processing signal (V1), which is configured to detect both stationary and non-stationary noise, wherein said analysis (12) has a lower frequency resolution than the frequency-bandwise noise reduction (21).and wherein, in the case of a detected presence (Y) of stationary and / or non-stationary noise in said analysis (12) of the processing signal (V1), the stationary noise is assumed to be detected in the frequency-bandwise noise suppression (21) in each frequency band (Bj, Bk), and the corresponding gain factors (Gk) are set.

The invention relates to a secondary harmonic current suppression method based on a fractional filter, which can realize secondary harmonic current suppression by designing a control loop of a preceding-stage DC / DC converter, does not need to use additional devices, and greatly reduces hardware cost and system complexity. The fractional order band-pass filter with the adjustable order is used for replacing a traditional integer order replacement filter, and the dynamic response speed of the system is effectively improved. The fractional order notch filter is introduced into the voltage control loop, the stability margin of the system can be guaranteed by adjusting the order, and in addition, the second harmonic current suppression effect is further improved by adjusting the gain coefficient of the fractional order notch filter. The problem that the system stability is reduced and the second harmonic current suppression performance improvement effect is limited due to a traditional integer order notch filter is solved.

The application provides a power management adaptive feedback regulation method and system with a wide voltage input range. In the application, the output current ripple waveform of the power supply is collected, wavelet packet decomposition is performed to extract the energy distribution of the characteristic frequency band, and the dynamic offset characteristics of the energy distribution with the wide voltage input fluctuation are used to identify the load change mode. The electromagnetic vibration caused by the current ripple is captured and converted into recovered electrical energy, which is directly supplied to the feedback circuit to realize energy self-supply. The recovered electrical energy is introduced into the ripple injection inverting proportional circuit, the gain coefficient and phase offset are dynamically set according to the load mode, and the inverting ripple signal is generated. The inverting ripple signal is superimposed on the feedback voltage to offset the ripple fluctuation by dynamically tracking the ripple amplitude change caused by the wide voltage input. Finally, the response speed of the rectifier tube driving timing is adjusted according to the input voltage fluctuation amplitude, and the optimized feedback control signal is generated. The application realizes the load fluctuation compensation and energy recovery closed loop.

Loudspeaker system comprising: a loudspeaker excursion characterization system (200; 300; 400) for a loudspeaker (126) with a signal generator (210) configured to generate a signal with several different amplitudes;and an inverse displacement filter (214) having an inverse displacement filter response, receiving the signal, applying the inverse displacement filter response, and having an output in communication with an amplifier circuit (112) of the loudspeaker (126), wherein the amplifier circuit (112) includes an input for receiving an output of the inverse displacement filter (214), a delay circuit (118) for receiving an input into the amplifier circuit (112), and a variable amplifier (122) having a variable gain factor and an input that receives an output of the delay circuit (118), wherein the inverse displacement filter response is an inverse of a displacement filter response of a displacement filter (114) in the amplifier circuit (112) of the loudspeaker (126).

The invention relates to the technical field of digital signal processing and power amplification, and discloses a power optimization method for high-power pulse amplification based on digital preprocessing. Pulse voltage is sampled and normalized into unit energy discrete pulse signals, frequency domain transformation is carried out under a unified energy reference, and frequency point amplitudes and phases are extracted to form a frequency spectrum sequence; establishing a frequency response model according to the cut-off frequency of the power amplifier to obtain an amplitude gain coefficient, constructing a pre-compensation function comprising an amplitude pre-compensation factor and a phase compensation amount changing along with the frequency, acting on a frequency spectrum sequence, and performing inverse transformation to form a pre-compensation time domain signal; after a power amplifier is input, an output frequency spectrum is reconstructed and inversely transformed to obtain an output time domain signal, peak power and frequency spectrum energy fidelity are calculated, power normalization is performed when the fidelity meets a threshold value, and a transmitting system is accessed, so that the problems of difficult compensation unification and frequency spectrum distortion are solved, and target power output is realized on the premise of ensuring frequency spectrum fidelity.

The invention discloses a device and a method for adjusting PLL (Phase Locked Loop) loop bandwidth based on VCO (Voltage Controlled Oscillator) working voltage. The device comprises a phase frequency detector, a digital filter, a digital oscillator and a frequency divider, the phase frequency detector is used for comparing the input signal with the output signal to obtain an error signal; the digital filter is used for processing the error signal; the digital filter comprises a proportional path; the digital oscillator is used for generating an output signal based on the processed error signal; the digital oscillator comprises a switch variable capacitor array; the switch variable capacitor array is used for adjusting the working voltage of the digital oscillator based on the error signal; the proportion path is directly connected with the phase frequency detector and the switch variable capacitor array; the frequency divider is used for carrying out frequency division on the output signal and feeding back the output signal to the phase frequency detector; by modifying the gain coefficient of the proportional path in the phase-locked loop, the adjustment of the loop bandwidth is realized, and the adjustment flexibility is improved.

The invention relates to the technical field of voice recognition, in particular to an audio device intelligent control method and system based on plant electric signals. The method comprises the following steps: acquiring a voltage signal, noise intensity and different environmental parameter values, and obtaining resting potential reference values and respective reference values; acquiring potential deviation and signal fluctuation amplitude based on the fluctuation amplitude of the voltage deviation in the time window and the overall condition; acquiring an environment gain factor and a noise suppression factor based on the difference between the environment parameter and the reference value and the difference between the noise and the reference value; pitch and tone quality are respectively adjusted through an environment gain factor, a noise suppression factor, potential deviation and signal fluctuation amplitude, and a special sound effect is determined in combination with voltage deviation; and music playing is controlled by combining musical instrument timbre recognition voice. According to the invention, the interactivity between the user and the plant is improved.

A fractional frequency divider configured to suppress spurious tones resulting from requantization of an accumulated quantization error has a multi-modulus divider (MMD) and one or more digital-to-time converters (DTCs). Each DTC includes a gain and / or digital predistortion, a dithered requantizer having a multibit input signa, a multibit output signal, and a multibit digitally-controllable delay (DCD). The quantized time delay between the input and output of the DTC is determined by the input from the MMD controller, the gain / digital predistortion and the requantizer. The MMD controller provides an input to the DTC that is proportional to the accumulated time quantization error introduced by the MMD controller, the gain / digital predistortion scales by a gain factor to produce a signal (a[k]) that is combined with a discrete-valued dither signal in the dithered requantizer to provide the control input of the DCD.

The invention relates to a phase adjusting circuit and method and a distance measuring device. The phase adjustment circuit comprises: at least one adjustment unit configured to adjust an input signal according to an initial gain coefficient to generate an adjustment signal; and changing a gain coefficient for adjusting the input signal in response to a gate control signal; wherein when the adjusting unit has different gain coefficients, the cut-off frequencies corresponding to the gain coefficients are the same; and the control unit is coupled with the adjusting unit and is configured to generate the gating control signal to the adjusting unit according to the adjusting signal and a preset signal. By adopting the scheme, the phase delay in the dynamic range can be reduced, and the distance measurement precision is improved.

A kind of DAS signal adaptive demodulation method and system based on FPGA, wherein, the method comprises: obtaining the in-phase component signal and quadrature component signal of DAS signal;When the address of write pointer occurs incremental change, the historical in-phase component signal and historical quadrature component signal in the preset history sampling window are read;The instantaneous amplitude energy of each time stamp is calculated, and the instantaneous amplitude energy sequence is obtained;The amplitude energy mean and variance eigenvalue of DAS signal are calculated;Adaptive gain factor is generated;The in-phase component signal and quadrature component signal are compensated and iteratively demodulated by adaptive gain, and the phase output value and demodulation module length value are obtained;The module length deviation amount of demodulation module length value and preset standard module length is calculated, and when the cumulative value is greater than preset locking threshold, the correction coefficient is calculated according to the current amplitude energy mean and module length deviation amount;The storage value of index address is updated based on correction coefficient.The application can improve the reliability of DAS system.

This invention discloses a sound source localization and speech enhancement method and apparatus. The speech enhancement method includes: converting dual-microphone signals to the frequency domain and converting the frequency domain signals to polar coordinates; calculating the slope range of the phase difference of the microphone signals based on the microphone spacing and sound speed, and dividing the phase difference slope according to accuracy requirements; converting each divided phase difference slope into phase compensation to the second microphone spectrum, and calculating the difference matrix and summation matrix; obtaining the best matching slope based on the difference matrix and summation matrix; expanding the best matching slope and combining it with the target sound source azimuth range to obtain the target slope interval; calculating the compensation spectrum of the second microphone based on the target slope interval, and performing target speech enhancement with the spectrum signal of the first microphone; calculating the gain factor by blocking the spectrum in the target direction and non-target direction; and further suppressing sound sources in non-target direction by comparing the relationship between the phase difference of the sound source and the phase difference in the target direction.

The embodiment of the invention provides an automatic gain control system and method based on a casing coupling positioning signal. The system is applied to the field of petroleum perforation engineering and comprises an analog front end, a prediction module, a master control module and a bus, and the analog front end comprises an over-temperature protection circuit, an over-current protection circuit, a level translation circuit, a buffer, a programmable amplification circuit and a filter circuit; wherein the level translation circuit is used for converting a bipolar signal into a unipolar signal; the buffer is used for isolating front-stage and rear-stage circuits; the programmable amplification circuit is used for adjusting a gain coefficient; the filter circuit is used for suppressing noise signals; the prediction module is used for pre-judging the voltage amplitude of the casing coupling positioning signal according to the real-time falling speed of the perforating gun; and the main control module is used for processing the falling speed data of the perforating gun, calling the prediction module to output a pre-judgment voltage value, matching an optimal gain coefficient according to the pre-judgment voltage value, and outputting a control instruction to the programmable amplification circuit through a bus, so that the quality of perforating data is improved.

A sample and hold circuit includes a pixel level connection coupled to a pixel cell. In a voltage domain global shutter operation, a reset voltage and a signal voltage are sampled and stored in a reset capacitor and a signal capacitor sequentially. In readout, the stored signal voltage is read out first to a comparing circuit to determine the best gain factor for the analog to digital converter (ADC) circuit to fully utilize the conversion range of the ADC to achieve a better signal to noise ratio. Once the better gain is determined and set for the ADC by comparing the signal voltage with a pre-determined threshold voltage, the same signal voltage is converted directly to become a digital signal value by the ADC. Then, the stored reset voltage is converted to a digital reset value by the ADC without changing the gain factor, to shorten the readout time.

Method for calibrating an evaluation circuit (200) for determining a resonant frequency (f0) of an electrical resonant circuit (100), wherein the evaluation circuit (200) comprises an evaluation coil (220), a transconductance amplifier (210) and a compensation circuit (230), wherein the evaluation coil (220) can be inductively coupled to the electrical resonant circuit (100), wherein an output of the transconductance amplifier (210) is connected via a feedback branch (260) to an input of the transconductance amplifier (210), wherein the evaluation coil (220) is arranged in the feedback branch (260) of the transconductance amplifier (210), wherein the compensation circuit (230) is arranged in parallel to the transconductance amplifier (210) and evaluation coil (220), the procedure comprises the following steps: - Interrupting the feedback branch (260) of the evaluation circuit (200); - Applying an alternating voltage (425) to the input of the transconductance amplifier (210); - Determining an output voltage (415) present at the output of the voltage amplifier (240); - Adjusting the gain factor (G) of the transconductance amplifier (210) to a value chosen such that the output voltage (415) at the output of the voltage amplifier (240) has a minimum amplitude; - Closing the feedback branch (260) of the evaluation circuit (200).

The invention discloses a method, device and system for detecting thermal runaway of a battery, a vehicle and equipment, and relates to the technical field of batteries. A specific embodiment of the method comprises the following steps: acquiring a current state of charge of a battery pack; according to the current charge state, determining a target gain multiple for amplifying an output signal of a pressure sensor and a pressure threshold value and / or a pressure change rate threshold value for thermal runaway judgment; according to the target gain multiple, amplifying an output signal of a pressure sensor in the battery pack to obtain an amplified signal; according to whether the amplified signal meets the pressure threshold value and / or the pressure change rate threshold value, whether thermal runaway happens to the battery or not is determined. According to the embodiment, thermal runaway of the battery can be detected in time.

The application discloses a speech processing method and device, a storage medium and a computer device. The method comprises the following steps: collecting an initial speech frame in a speech signal, obtaining an initial spectrum corresponding to the initial speech frame and a global speech existence probability, and obtaining an initial power spectrum based on the initial spectrum; determining each target frequency point of a noise increasing stage based on the initial power spectrum, and obtaining a local speech existence probability corresponding to each frequency point; performing probability correction on the local speech existence probability corresponding to each target frequency point based on the global speech existence probability, so as to obtain a target speech existence probability corresponding to each target frequency point; obtaining a gain factor corresponding to each frequency point based on the target speech existence probability corresponding to each target frequency point and the local speech existence probability corresponding to other frequency points; performing gain processing on the initial spectrum based on the gain factor corresponding to each frequency point, so as to obtain a target spectrum; and generating a target speech frame corresponding to the initial speech frame based on the target spectrum. The application improves the intelligibility of the target speech frame.

The present application relates to the technical field of intelligent decision-making, in particular to an engineering change intelligent identification and influence analysis method and system. In the present application, by extracting the physical connection information of the field control equipment and associating the operating parameters and the logical constraint parameters, a field device interconnection topology model with signal flow direction as the connection line is constructed, the discrete engineering equipment is converted into a digital map with logical handshake relationship, the parameter modification behavior is monitored in real time and the change source equipment is locked, the history trend consistency and real-time availability of the redundant signal channel are verified by using the grey correlation analysis algorithm and the effectiveness threshold, when it is confirmed that it has effective backup, it is automatically determined as a blocking node, the input signal change is simulated and calculated according to the PID control algorithm combined with the loop gain coefficient, whether the fluctuation occurs physical dissipation is confirmed by comparing the absolute value of the difference of the theoretical output signal and the signal control dead zone threshold, and the effective fluctuation which can actually drive the downstream equipment action is accurately identified.

The invention discloses a signal amplifier design method based on a spin bit-nano mechanical oscillator composite system. The signal amplifier design method comprises the following steps: 1) constructing the spin bit-nano mechanical oscillator composite system; 2) using a strong pumping field, a weak detection field and a magnetic field to jointly act on the spin bit-nano mechanical oscillator composite system in the step 1) to realize coupling of spin bits and nano mechanical oscillators; and 3) adjusting the intensity of the strong pumping field in the step 2) according to the gain factor alpha of the expected detection wave. According to the signal amplifier design method based on the spin bit-nano mechanical oscillator composite system, the design of the signal amplifier based on the spin bit-nano mechanical oscillator composite system can be guided; according to the invention, the amplification factor of the signal amplifier can be adjusted by changing the intensity of the strong pumping field.

The invention discloses a hybrid intelligent reflector system based on FPGA centralized control. The system comprises a reflective array layer, a control layer, a communication feedback layer and an energy management module. The reflective array layer reflects the incident wave signal to a target direction through active amplification and phase adjustment; the control layer adopts an FPGA centralized control structure, establishes a channel model based on channel state information, executes a beam forming algorithm and phase matrix calculation, generates a target phase control matrix and a power distribution scheme, and adjusts a gain coefficient of the reflective array layer; the communication feedback layer extracts channel estimation parameters and carries out channel data reconstruction; and the energy management module transmits a real-time load signal of the reflective array layer to the control layer, receives a control level signal of the control layer, and cooperatively adjusts the power and phase of the reflective array layer. According to the invention, on the basis of a centralized control architecture and an energy adaptive management mechanism of the FPGA, signal amplification compensation, intelligent beam direction regulation and control and dynamic power distribution are realized.