811 results about "Equalizer" patented technology

The invention discloses a time-frequency synchronization tracking and judging method based on GFSK (Gaussian Frequency Shift Keying). The method comprises the following steps: S1, positioning the initial position of a synchronization word by detecting a lead code; s2, extracting a fixed synchronization word, completing frame synchronization based on correlation matching, and positioning a starting point of an effective data segment; s3, a range cycle counter collects a central sampling point and two adjacent sampling points to form a candidate sampling point sequence; s4, dynamically and finely adjusting the range cycle counter to form a time sequence calibration sampling point sequence; s5, performing phase difference calculation on the time sequence calibration sampling point sequence, estimating continuous frequency offset and generating a compensation value; s6, applying the compensation value to construct a decision feedback equalizer with delay front and back terms; s7, performing interference suppression on the compensation sampling point sequence by using a decision feedback equalizer with delay front and back items, and outputting a soft decision value; and S8, executing three times of judgment, constructing a sliding reference sequence, and outputting a final bit judgment result. According to the method, the problem of poor GFSK demodulation robustness under Doppler disturbance is solved.

The invention discloses a dual-mode analog front end with a full-speed clock-free decision feedback equalizer (UC-DFE). The dual-mode analog front end comprises a terminating impedance matching network, a linear processing link, the UC-DFE and a signal output module, the terminating impedance matching network realizes broadband impedance matching and electrostatic protection; the linear processing link comprises a CTLE and a VGA and is used for compensating channel attenuation and expanding signal bandwidth; the UC-DFE comprises a summator, a static comparator, a delay unit and a feedback tap, and realizes high-speed signal processing and feedback equalization; and the signal output module converts the balanced signal into a target coding format and outputs the signal. According to the invention, the overall performance of the circuit is improved, the problems of time sequence tension, bandwidth limitation and the like of the traditional DFE architecture at a high speed are solved, and the circuit is suitable for 128Gbps high-speed PAM4 and NRZ signal transmission and shows better comprehensive performance.

The invention discloses a self-adaptive suppression method and system for multi-band signal interference of a mobile terminal, and the method comprises the steps: collecting a multi-band radio-frequency signal power spectrum, extracting an interference feature vector, inputting the interference feature vector into a neural network prediction model, outputting future interference intensity probability distribution, generating a prediction frequency spectrum according to the probability distribution, and combining the current channel state information, pre-compensation processing is carried out through a double-level equalizer; the balancing strategy is dynamically adjusted according to the resource state of the mobile terminal, specifically, when the CPU load exceeds a first preset value or the electric quantity is lower than a second preset value, a second-level equalizer in the double-level equalizer is forbidden, and the steps S20-S40 are executed again till the CPU load and the electric quantity meet preset conditions; and a sliding window verification mechanism is adopted, DTW distances of the predicted frequency spectrum and the actually measured frequency spectrum are compared, and the frequency band weight coefficient of the feature extraction matrix is dynamically updated. According to the method, the signal interference suppression capability is improved through the synergistic effect of time-frequency feature enhancement, hybrid model prediction, dynamic resource scheduling and closed-loop optimization.

The invention relates to the technical field of distributed response, in particular to a distributed message dynamic response method based on websocket. The method comprises the following steps: constructing a distributed message service cluster to configure a websocket communication module, message middleware and a client for each service node, and providing a unified service access point to the outside through an Nginx load balancer, so as to automatically register and subscribe a global control theme to the message middleware and form a hierarchical theme subscription structure; when the client initiates a connection request, the connection is distributed and a secure channel is established; the client sends a service request message, packages the service request message into a standardized message format, and associates session information of the client at the same time; and receiving the session message, performing intelligent filtering, generating a response result, and pushing the response result to the client in real time. According to the invention, efficient processing and real-time response of messages in a distributed environment can be realized.

The invention relates to the technical field of artificial intelligence and digital audio signal processing, and particularly discloses a Karaoke all-in-one machine intelligent tuning method based on AI sound effect optimization, and the method comprises the steps: obtaining an original mixed audio signal sung by a user, and carrying out the preprocessing; processing the preprocessed audio by using a multi-source audio separation model based on a deep neural network, and extracting a human voice track and accompaniment components; extracting time-frequency domain features from the human voice track, and performing quantitative evaluation on the voice tension degree of the user in combination with the trained voice state recognition model; dynamically adjusting an equalizer and a dynamic range compression parameter according to an evaluation result to realize adaptive sound effect optimization; synthesizing the optimized voice and accompaniment and outputting the synthesized voice and accompaniment to monitoring equipment; and finally, establishing a personalized user behavior feature vector by collecting user singing performance and historical tuning data, and performing online fine tuning on a system strategy based on a deep Q network reinforcement learning mechanism to continuously optimize a tuning effect.

The invention discloses a debugging method and system of an audio system, and relates to the technical field of audio debugging, the method comprises the steps of S1, environmental acoustic analysis, S2, basic parameter presetting, S3, real-time monitoring and dynamic adjustment, and S4, adaptive optimizing.The layout of an equalizer or a loudspeaker is adjusted through environmental acoustic analysis; presetting basic parameters of the sound amplifying device and the microphone; real-time multi-band sampling, delay calibration and gain regulation are realized; adaptive optimization is carried out, algorithms such as noise reduction are started, stored parameters are scene presetting, and one-key calling and self-checking updating are supported; the environment is analyzed, the audio system is accurately debugged, the system is dynamically optimized, the tone quality and the debugging efficiency are improved, and diversified scene requirements are met.

This disclosure relates to a receiver device for pulse amplitude modulation (PAM) signals. The receiver device calculates a transmitter dispersion eye closure quaternary (TDECQ). The receiver device first obtains a signal, wherein the signal is based on a PAM signal sent by a transmitter device over a channel to the receiver device, and filters the obtained signal. Further, the transmitter device equalizes the filtered signal using a FFE with multiple taps, and filters the equalized signal output by the FFE using a 2-tap post filter, wherein high frequency noise caused by the FFE is compressed. The receiver device applies a Max-Log-Map (MLM) algorithm on the filtered signal output by the 2-tap post filter, reconstructs a signal constellation of the PAM signal based on the result of the MLM algorithm, and calculates a TDECQ based on the reconstructed signal constellation.

The utility model provides an equalizer adjustable circuit, an amplifier adjustable circuit and electronic equipment, and belongs to the technical field of equalizers. And the cross negative impedance module is used for compensating the input signal through negative impedance. And the adjustable current module is connected with the cross negative impedance module. And the adjustable current module is used for providing adjustable bias current for the cross negative impedance module so as to adjust the negative impedance generated by the cross negative impedance module through the adjustable bias current. By adjusting the current of the adjustable current module, the bias current of the cross negative impedance module can be adjusted, and then the negative impedance generated by the cross negative impedance module can be adjusted, so that the loss of the circuit and the frequency response characteristic can be optimized in bandwidth, gain, peak point and peak amplitude; the technical problem that a traditional equalizer circuit cannot adjust the circuit gain is solved, so that the circuit can achieve the equalization effect, and input signals are compensated and equalized.

The invention discloses a high-speed serial signal linear equalization circuit which comprises an input terminal, a continuous time linear equalizer (CTLE), a super source follower, a variable gain amplifier (VGA) and an adaptive feedback. The input end is connected with a 100-ohm internal resistance differential signal source of the transmitting end through a channel; the equalization circuit CTLE is connected to the output end of the input end connection module, two groups of resistance switches (RsMF and RsHF) of the CTLE module are respectively controlled by 4-bit binary codes to enable source degeneration resistors Rs of two stages of CTLEs to be variable, two groups of capacitance switches (CsMF and CsHF) are respectively controlled by 2-bit binary codes to enable source degeneration capacitors Cs of the two stages of CTLEs to be variable, and the two groups of capacitance switches (CsMF and CsHF) are respectively controlled by 2-bit binary codes to enable source degeneration capacitors Cs of the two stages of CTLEs to be variable. The change of a zero pole of the circuit is realized through the change of the source degeneration resistor Rs and the source degeneration capacitor Cs, and the adjustability of the difference between the high-frequency gain and the low-frequency gain of the circuit is realized, so that the multi-gear adjustability of the equalization capability is realized.

An equalizer includes a delay circuit configured to generate at least one delay signal based on at least one input signal, an encoding circuit configured to generate a plurality of driving signals and a connection switching signal, based on the input signal and the delay signal, a pre-driver configured to generate a plurality of driving switching signals based on the plurality of driving signals, a first differential circuit configured to generate a first differential output signal based on the plurality of driving switching signals, a second differential circuit configured to generate a second differential output signal based on the plurality of driving switching signals, and a connection circuit configured to adjust a connection between the first differential circuit and the second differential circuit based on the connection switching signal.

The invention discloses an LPO wired receiver adopting adaptive CTLE based on a genetic algorithm, and the receiver comprises an analog front end, a data sampler, an edge sampler, an error sampler, a 1: 32 demultiplexer, a verification module, an eye pattern monitor, a genetic algorithm module, clock recovery logic, and a clock path. The eye pattern monitor is used for calculating the eye pattern closure degree according to the data of the data sampler and the error of the error sampler and outputting the eye pattern closure degree to the genetic algorithm module; and the genetic algorithm module is used for realizing parameter adaptive control of a continuous time linear equalizer CTLE and a variable gain amplifier VGA in the analog front end. The invention aims to apply the genetic algorithm to the adaptive equalization of the adaptive continuous time linear equalizer, so that the adaptive continuous time linear equalizer has stronger global search capability and lower power consumption and is not easy to fall into local optimum, and the complexity and power consumption of the adaptive continuous time linear equalizer are reduced. And the real-time and energy-saving requirements are met.

The invention discloses a temperature compensation equalizer with adjustable equalization slope linearity, and relates to the technical field of microwave integrated circuits. The equalizer comprises a temperature compensation driving module and a temperature compensation equalization module. The temperature compensation driving module generates a control voltage changing along with the temperature by using the negative temperature characteristic of the diode. The temperature compensation equalization module comprises an equalization circuit formed by resonators and a pair of stacked switch tubes, and realizes a temperature compensation function by controlling voltage to adjust insertion loss; meanwhile, by changing the capacitance value of the equalization slope linearity adjusting capacitor C3 which is connected with the series resonator in parallel, the linearity of the equalization slope can be flexibly adjusted on the premise that the equalization quantity is kept unchanged. According to the invention, the temperature compensation and equalization functions are integrated, the structure is compact, the performance is stable, the slope linearity can be flexibly adjusted, and the device is suitable for gain frequency and temperature compensation in a broadband radio frequency link.

The invention provides a continuous time linear equalizer based on a low-pass and band-pass mixed structure. The continuous time linear equalizer comprises a frequency selection coupler, a low-pass transconductance unit, a band-pass transconductance unit and a load unit. The frequency selection coupler is used for separating the differential analog signal into a first voltage component in a first band-pass frequency range and a second voltage component in a low-pass frequency range, the first band-pass frequency range comprises a target frequency expected to be compensated, and the frequency in the first band-pass frequency range is greater than the frequency in the low-pass frequency range; the band-pass transconductance unit is used for converting the first voltage component into current and synchronously realizing gain amplification to obtain a first current signal; the low-pass transconductance unit is used for converting the second voltage component into current and synchronously realizing gain amplification to obtain a second current signal; the load unit is used for summing the first current signal and the second current signal and converting the sum to a differential output voltage signal. According to the invention, the frequency corresponding to the gain peak point and the gain amplitude can be improved.

A receiver device receives a signal transmitted to the receiver device over an optical communication channel and equalizes the signal using a multi-decision feedback equalizer of the receiver device. Equalizing the signal includes generating, using at least one decision feedback equalizer configured with a plurality of slicing thresholds, decisions on symbols transmitted to the receiver device, detecting that a decision made by the decision feedback equalizer is unreliable. Equalizing the signal also includes, in response to detecting that the decision is unreliable, tracking, for a tracking period, multiple decision paths that generate respective possible sequences of symbols transmitted to the receiver device, determining error energies in decisions made, during the tracking period, in respective decision paths, and selecting, based on a comparison between the respective error energies, a sequence of symbols generated in one of the multiple decision paths as an output of the multi-decision feedback equalizer.

Serializer / deserializer (SerDes) modules, equalizers, and equalization techniques having robust parameter initialization may substantially reduce convergence time. One illustrative equalizer includes: a discrete-time, finite impulse response (“FIR”) filter to convert a receive signal to a filtered signal; a decision element to determine channel symbols represented by the filtered signal; and a controller. The controller is configured to, for each of multiple values in a search range, estimate a performance of the equalizer based on the channel symbols and at least one of the filtered signal or an input signal to the decision element; configured to find a centroid value based on the performance for each of the multiple parameter values in the search range; and configured to derive an initial value for the parameter from the centroid value.

The invention provides an optical module and a link optimization method and system. The method comprises the following steps that: the microcontroller issues a gain value, so that the continuous time linear equalizer performs equalization processing on an input voltage signal; the time processing module is used for sampling a voltage signal output by the continuous time linear equalizer to obtain sampling data; and the microcontroller determines the gain configuration of the continuous time linear equalizer based on the sampling data under different gain values. Compared with the prior art, the power consumption and the time delay of the module are equal to those of a traditional LPO, and meanwhile, the self-adaptive capability which is only achieved by a DSP scheme in the past is obtained. The application scene and deployment flexibility of the LPO technology are expanded; and meanwhile, the link performance can be always maintained near the optimal point, so that the risk that the performance is continuously deteriorated until error codes occur is fundamentally avoided, and intelligent maintenance of the link state is realized.

Systems and methods are provided for a memory device that includes a decision feedback equalizer (DFE) reset generator configured to transmit a DFE reset signal to reset taps of a DFE. The memory device also includes an input buffer. The input buffer includes a data branch configured to output data from the input buffer for use downstream in the memory device. The input buffer also includes a DFE reset branch configured to reset the taps for the DFE based on the DFE reset signal. Moreover, resetting the taps using the DFE reset branch does not reset output data of the data branch.

The invention discloses a self-adaptive CTLE gain adjustment method and system in a high-speed digital transmission system, relates to the technical field of equalizers of high-speed digital signal transmission systems, and realizes self-adaptive adjustment of CTLE by dividing a state interval of a first tap coefficient of a DFE and analyzing a transfer trend of the state interval. The method specifically comprises the following steps: dividing a first tap coefficient of the DFE into four states according to three set boundary reference lines; sampling a first tap coefficient of the DFE in a signal processing period and recording a current state of the first tap coefficient; and judging the validity of the state transition by checking the consistency of the previous two states, if the state transition is valid, determining a gain adjustment strategy according to the state transition condition, and adjusting the CTLE gain according to the gain adjustment strategy, otherwise, maintaining the current gain. According to the invention, through interval dynamic feedback of the first tap coefficient of the DFE, stable convergence of the CTLE gain is realized, and intersymbol interference and noise influence are effectively suppressed.

The invention belongs to the technical field of electronic information, and discloses a millimeter wave magnetic coupling variable resonant equalizer, which comprises a transmission waveguide channel, a coupling channel and a resonant cavity which are communicated, the resonant cavity and the coupling channel are arranged in the narrow side direction of the transmission waveguide channel, the opening direction of the coupling channel is parallel to the direction component of an electric field, and the opening direction of the coupling channel is parallel to the direction component of the electric field. The inner walls of the two ends of the resonant cavity are adjustably and slidably connected with adjusting blocks, the opposite inner side faces of the two adjusting blocks are fixedly connected with wave absorbing materials, and the inner wall, facing the coupling channel, of the resonant cavity is detachably and movably connected with a guide block. The center position of the resonant cavity is provided with a telescopic tuning pin perpendicular to the direction component of the magnetic field. According to the structure, the processing and installation problems of a common waveguide equalizer are effectively solved, the equalizer is debugged more simply and quickly, meanwhile, the standing-wave ratio is optimized by changing the cavity structure, and a relatively complex gain curve can be equalized through the good frequency response characteristic.

A virtual radio frequency (VRF) equalizer for an envelope tracking integrated circuit (ETIC) is disclosed. In one aspect, an ETIC provides envelope tracking (ET) for a barely Doherty (BD) power amplifier stage. The VRF equalizer includes circuitry that provides ripple cancelation that is caused by load modulation of the BD power amplifier stage. Additional circuitry is included to compensate for an amplifier within the ETIC. By canceling the ripple within the ETIC, the overall performance and efficiency of the BD power amplifier stage is improved, resulting in better performance of a transmitter in a wireless communication device.

Data transceivers including a transmitter configured with a digital filter, and a receiver configured with an equalizer, are equipped with logic to measure a combined pulse response to a symbol generated by the transmitter at an output of the equalizer, adjust the combined pulse response by an impulse response of the digital filter to generate an adjusted pulse response, and modify the digital filter with one or more floating taps based on the adjusted pulse response.

In one embodiment, a method is disclosed for port reduction, comprising: calculating, at a distributed unit (DU), beamforming weights based on received DMRS channel estimates; sending the calculated beamforming weights to a radio unit (RU); applying, at the RU, port reduction to DMRS symbols based on the calculated beamforming weights; compressing IQ data and received DMRS symbol data to compensate for an underlying bandwidth increase, thereby enabling beamforming at the RU. PUSCH data may be compressed. Beamforming weight calculations may be performed in the O-DU. Beamforming weight calculations may be provided to the L2 scheduler. BFP format compression may be performed. Channel information may be compressed for non-signal data. DMRS compression may be performed at the CU. The DMRS pattern may be received from the control plane. An equalizer weight calculation may be performed to merge a plurality of CSI estimates from different RUs for a given user to calculate joint equalization. The method may further comprise applying, at the RU, port reduction to both PUSCH and DMRS symbols. Beamforming may be performed for uplink, for downlink, or both.

The application discloses a loudspeaker phase equalization signal processing system and a pole parameter determination method thereof, which equalizes the loudspeaker phase response in a set equalization frequency range to linear phase, and the system comprises a phase controller and a group delay equalizer in cascade; the phase controller is a first-order IIR all-pass filter; the group delay equalizer comprises S low-order IIR all-pass filters in cascade, wherein the first N1 low-order IIR all-pass filters are low-order causal IIR all-pass filters, and the last N2 low-order IIR all-pass filters are low-order non-causal IIR all-pass filters. The application can equalize the loudspeaker phase through the all-pass filter with low operation complexity and parameters, can significantly reduce the calculation amount required by low-frequency phase equalization, can improve the linear degree of the loudspeaker phase response, and can improve the sound quality and listening experience.

Apparatuses and methods in a communication system are disclosed. A signal is received. The signal comprises demodulation reference information. The apparatus compute estimates of the channel and of interference-plus-noise vectors. The interference-plus-noise vectors are used as input for choosing a tapering matrix and for determining shrinkage target matrix and parameter. An interference plus noise covariance matrix estimator is computed and used in an equalizer, which has as input an estimate of the interference-plus-noise covariance matrix.

An example apparatus includes: equalizer circuitry having first and second inputs and an output, the equalizer circuitry configured to adjust an audio signal at the first input responsive to coefficient values at the second input; amplifier circuitry having an input coupled to the output of the equalizer circuitry, and having an output; control circuitry having a first input coupled to the first input of the equalizer circuitry, having a second input coupled to the output of the amplifier circuitry, and having an output coupled to the second input of the equalizer circuitry, the control circuitry configured to determine a resonant frequency of a speaker device responsive to the audio signal and a signal at the output of the amplifier circuitry, and to provide the coefficient values at the output of the control circuitry responsive to the resonant frequency of the speaker.

The invention provides a self-adaptive equalization circuit and a self-adaptive equalization method, and relates to the technical field of data communication. The adaptive equalization circuit comprises an equalizer, a gain amplifier, a signal acquisition unit and a digital control unit, the signal acquisition unit comprises a low-pass filter, a first power detection module and a first comparator, and the output end of the equalizer is connected with the gain amplifier to form a signal path. The output end of the equalizer, the low-pass filter, the first power detection module and the first input end of the first comparator are sequentially connected, the second input end of the first comparator is used for inputting a reference level, and the digital control unit is respectively connected with the output end of the first comparator and the gain amplifier; the digital control unit is used for adjusting the reference level according to the first comparison result until the first comparison result output by the first comparator conforms to a preset signal mode; and the gain of the gain amplifier is adjusted according to the corresponding reference level in the preset signal mode. The method has the effect of reducing power consumption.

The invention discloses a method for designing a double-iteration Turbo equalizer of an OTFS-MIMO-ISAC system, and the method comprises the steps: carrying out the sparse processing of covariance matrix inversion, and reducing the matrix inversion cost of conventional MMSE equalization through linear complexity derivation; the sparse linear system is solved through the TFQMR algorithm to reduce the complexity of the first iteration, and a sparse approximate inverse matrix is generated in the subsequent iteration through the FSPAI algorithm to further realize the balance between the calculation efficiency and the performance; and dynamically optimizing channel information by adopting an iterative channel estimation method and utilizing MMSE (Minimum Mean Square Error) equilibrium feedback. Through outer iteration and inner iteration, the symbol estimation precision is gradually improved, meanwhile, the calculation complexity is reduced, and a low-complexity and high-precision equalization and channel estimation solution is provided for a high-mobility wireless communication system.

A method for equalizing a data signal includes providing a data signal to a pair of input transistors that provides an output signal, contributing to the output signal, a first current or voltage that has a magnitude determined by a first weighted feedback signal using a first decision-feedback circuit that includes a first pullup circuit coupled to a drain of a first input transistor, and a first pulldown circuit coupled to the drain of the first input transistor, and contributing to the output signal, a second current or voltage that has a magnitude determined by a second weighted feedback signal using a second decision-feedback circuit that includes a second pullup circuit coupled between a drain of a second input transistor and the first voltage rail, and a second pulldown circuit coupled between the drain of the second input transistor and a second voltage rail.

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 an unmanned aerial vehicle electromagnetic interference positioning method and system, and relates to the technical field of unmanned aerial vehicles, and the method comprises the steps: 1, carrying out the broadband frequency spectrum scanning of a remote control link frequency band, a telemetering link frequency band and an image transmission frequency band at the full flight stage of an unmanned aerial vehicle, detecting an interference signal in a frequency hopping communication system in real time, and transmitting the interference signal to an unmanned aerial vehicle; identifying random hopping frequency conversion points covered by the interference signals to form a hopping frequency conversion point set; and step 2, inputting the hopping frequency point set into a radio frequency sharing front end, coupling a radio frequency signal of a communication antenna to a spectrum analysis unit through a duplexer, and performing signal compensation on a coupling path by utilizing automatic gain control and a phase equalizer to obtain corrected interference spectrum data. The anti-interference capability and the flight safety of the unmanned aerial vehicle in a complex electromagnetic environment are improved, and the avoidance path is generated to guarantee the safe flight of the unmanned aerial vehicle.