80 results about "Gain factor" patented technology

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 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 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.

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.

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.

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 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.

An interference signal detection method, a storage medium and an electronic device are applied to a baseband signal processing process of a navigation receiver, and the method comprises the following steps: obtaining an average power value of an intermediate frequency signal in a preset window after the intermediate frequency signal is processed by a digital AGC module; according to the average power value of each window, determining a target gain coefficient of the digital AGC module, so that the average power value meets a preset power stability condition in a plurality of continuous windows; determining whether a strong power interference signal exists in the intermediate frequency signal according to a preset reference gain coefficient and the target gain coefficient; wherein the reference gain coefficient is a gain coefficient of the digital AGC module in a time period when no deception interference is confirmed, and the detection of a strong power interference signal is realized.

An image sensor, an analog-to-digital converter, and a method of performing readout of an image sensor pixel using the same. The ADC may include a sample and hold circuit having a plurality of sample and hold branches, where each branch is coupled to a separate DAC. A total capacitor area in the ADC for all the individual DACs may sum to the total capacitor area required in a non-branched ADC configuration. An exemplary ADC may enable analog correlation multi-sampling to reduce noise or otherwise improve signal-to-noise ratio. The ADC may include a configurable gain factor and a configurable number of correlated multi-sample reads. Exemplary embodiments include an image sensor having a pixel output coupled with a readout circuit, where the readout circuit includes an ADC having a split DAC architecture, enabling lower read noise and higher frame rates at lower power consumption.

A buzzer driving system is electrically connected to a buzzer and includes a driving unit, a boost unit, and a gain estimation unit. The driving unit includes a gain control circuit that adjusts the gain of an audio signal based on a gain factor and outputs a driving voltage to the buzzer based on the adjusted audio signal. The boost unit is electrically connected to the driving unit and boosts an input voltage to generate a working voltage for the driving unit. The gain estimation unit is electrically connected to both the driving unit and the boost unit and is configured to generate a plurality of stepwise voltages based on the working voltage. The gain estimation unit sequentially compares the input voltage with each of the stepwise voltages to generate an estimated gain. The gain factor used by the gain control circuit is determined based on this estimated gain.

The present disclosure discloses an array pickup method and apparatus capable of steplessly adjusting a pickup direction. The method includes: receiving a first audio from a unidirectional mic capsule and a second audio from a nondirectional mic capsule, and performing calculation to obtain a gain coefficient; performing gain processing on the first audio according to the gain coefficient to obtain a first gain audio; performing delay alignment on the second audio to obtain an auxiliary audio; performing sound mixing processing on the first gain audio and the auxiliary audio according to a proportion value to obtain a sound-mixed audio; and performing noise reduction processing on the sound-mixed audio according to a noise reduction strategy and the gain coefficient to obtain a corresponding target audio. A noise reduction function stronger than that of traditional single mic capsule or simple array pickup is achieved.

The invention discloses a carrier tracking method, system, equipment and medium for a second-order frequency-locked loop assisted third-order phase-locked loop, and the method comprises the steps: receiving an in-phase integral value and a quadrature integral value outputted by a correlator, and carrying out the second-order frequency-locked loop tracking and the third-order phase-locked loop tracking based on the in-phase integral value and the quadrature integral value, the control quantity of the frequency-locked loop and the phase-locked loop is obtained; estimating a carrier-to-noise ratio and a carrier wave attitude of the received signal corresponding to the in-phase integral value and the orthogonal integral value in real time, and determining a first gain coefficient and a second gain coefficient according to the carrier-to-noise ratio and the carrier wave attitude; and generating a final control quantity of the carrier numerical control oscillator based on the control quantities of the frequency-locked loop and the phase-locked loop, the first gain coefficient and the second gain coefficient. By introducing a real-time sensing and dynamic gain control mechanism based on the carrier-to-noise ratio and the carrier wave fluctuation attitude, weighted fusion of contribution of the frequency-locked loop and the phase-locked loop is realized, and the robustness, the precision and the adaptive capacity of the carrier tracking loop under all working conditions can be improved.

The invention provides a millimeter wave radar signal processing method and device, a storage medium and a computer program product, and relates to the technical field of radar signal processing. The method comprises the following steps: framing and windowing an original time domain signal; performing fast Fourier transform on the windowed time domain signal to generate an initial frequency domain signal; determining a dynamic gain factor according to the signal-to-noise ratio of the current frame; determining a smoothing factor according to the dynamic gain factor; performing smoothing processing on the dynamic gain factor by using the smoothing factor and the smoothed gain factor of the adjacent frame to generate a smoothed gain factor of the current frame; generating a frequency spectrum parameter subjected to smoothing gain according to the smoothed gain factor and the initial frequency spectrum parameter; performing inverse fast Fourier transform on the frequency domain signal subjected to smooth gain processing to obtain an initial de-noised time domain signal; and windowing and overlapping adding are carried out on the frame of the initial denoising time domain signal to generate a final denoising time domain signal. The present invention can suppress music noise.

The application provides a vehicle sound real-time synthesis method and system based on multi-track dynamic interpolation, comprising the following steps: S1, determining a synthesis target, collecting and analyzing multi-track original audio sample data; S2, determining a minimum frame sampling point number, respectively cutting sample units in each track audio in combination with a dynamic pitch change rate, and performing fast linear interpolation on the single-frame audio sampling points after cutting of each track; S3, calculating a dynamic gain coefficient of each track audio according to a real-time working condition of a vehicle, and performing gain processing on the single-frame sample after interpolation; and S4, synthesizing single-track audio from the gain-processed audio of each track according to a preset weight, and frame-by-frame splicing the synthesized single-track audio to synthesize final audio. The application also provides an in-vehicle acoustic implementation device, comprising an audio storage module, a working condition parameter collection module, a data processing module and an audio output module, which are used for realizing output of target sound audio and realizing real-time playing.

Provided is an electronic device including a first signal line configured to receive a first reception signal, a second signal line configured to receive a second reception signal, a third signal line configured to receive a third reception signal having a correlation with the second reception signal, the third signal line being positioned between the first signal line and the second signal line, a first crosstalk filter connected to the second signal line and configured to output a first compensation signal by applying a pole frequency and a gain coefficient to a second reception signal, and an adaptive filter configured to receive the second reception signal and a first final signal corresponding to the first reception signal, and to adjust the pole frequency and the gain coefficient of the first crosstalk filter based on the second reception signal and the first final signal.

An input device comprises a touch input element installed on a vehicle's steering wheel and supported in such a way that it can vibrate, a detection element to recognize an input operation on the touch input element, a vibration output element to vibrate the touch input element, and a controller to actuate the vibration output element, wherein, when the input operation is recognized by the detection element, the controller causes the touch input element to vibrate by actuating the vibration output element with a first signal, a vibration transmission element is provided on a vibration transmission path from the touch input element to the steering wheel, and the first signal actuates the vibration output element at a frequency in a vibration damping range in which a resonance gain factor of a vibration system provided on the vibration transmission path is less than zero.

The invention discloses an automatic test system and method for a four-quadrant positioning processing module. The automatic test system comprises a control unit, a signal generation unit, a power supply, a gain coefficient test module, a shaping function test module and a smooth direct current output test module, stable micro-signals are transmitted to the four-quadrant positioning processing module and the controllable gain coefficient testing module through the signal sounding unit, and the controllable gain coefficient testing module, the single-side shaping function testing module, the smooth direct current output testing module and the controller are used for testing the stable micro-signals. Accurate calculation of a controllable gain coefficient, a single-side shaping function and smooth direct current output of the four-quadrant positioning processing module can be realized.

Dynamic range control (DRC) cannot be simply applied to higher-order ambisonics (HOA) based signals. [Solution] A method for performing DRC on an HOA signal includes converting the HOA signal to the spatial domain, analyzing the converted HOA signal, and obtaining a gain factor usable for dynamic compression from the results of the analysis. The gain factor can be transmitted together with the HOA signal. When DRC is applied, the HOA signal is converted to the spatial domain, the gain factor is extracted, and in the spatial domain, it is multiplied with the converted HOA signal to obtain a gain-compensated converted HOA signal. The gain-compensated converted HOA signal is converted back to the original HOA domain to obtain a gain-compensated HOA signal.