208 results about "Equaliser" patented technology

The invention discloses a parameter configuration method and device of an equalizer and frame-type equipment. The method comprises the following steps: acquiring a transmission characteristic corresponding to a first transmission signal; according to a transmission characteristic corresponding to the first transmission signal, updating an initial parameter of the target equalizer to obtain a first adjustment parameter; adopting a mode of obtaining a first adjustment parameter based on the initial parameter to carry out iteration updating on the parameter of the target equalizer for multiple times to obtain a plurality of adjustment parameters and transmission characteristics of the transmission signals corresponding to the plurality of adjustment parameters, and determining a change trend of the transmission characteristics of the transmission signals corresponding to the plurality of adjustment parameters; determining a parameter corresponding to the transmission characteristic of which the change trend is changed for the first time as a target parameter of a target equalizer; and configuring the target equalizer according to the target parameter. According to the invention, the technical problem of low efficiency of the parameter configuration method of the equalizer in the prior art is solved.

The invention discloses a high-concurrency real-time message pushing system and method based on an SSE protocol, and relates to the technical field of communication. Comprising the steps of 1, establishing a high-concurrency real-time message pushing system, 2, performing access, heartbeat detection, connection state management and load balancing of a client through an access module, performing dynamic strategy allocation based on tenant ID hash through a load balancer of the access module, routing client requests to different SSE servers, and sending the client requests to the SSE servers. 3, caching and routing management of messages are carried out through a message management module, distributed message middleware kafka is used as a cache queue through the message management module, real-time messages sent by a service system are received, and the messages are stored in a classified mode according to top and Tenant; the connection metadata comprises a connection ID, a client subscription topic and a timestamp, and step 5, monitoring the connection number of each node in real time through a monitoring module, triggering connection migration and message reissuing when the node fails, removing the information of the failed node, and notifying a load balancer to route the original connection to other nodes. And meanwhile, the message management module is notified to re-push the message which is not successfully pushed by the fault node to the new node, so that the message is prevented from being lost.

The invention relates to the technical field of audio processing, in particular to a method for automatically generating parameters of an audio equalizer based on deep learning. The method comprises the following steps: collecting multi-source audio data and carrying out short-time Fourier transform and principal component analysis, generating a dimension reduction feature set, obtaining reference template data and constructing a parameter space model, carrying out corresponding relation analysis on the dimension reduction feature set, generating mapping relation data, and obtaining the multi-source audio data. After a deep learning network is built based on mapping relation data and interlayer links are optimized, optimization model structure data are generated and new audio samples are collected, iterative training is carried out by utilizing an optimization model to generate a deep audio equalization parameter model, a candidate equalizer parameter set is predicted, the candidate parameter set is processed through difference correction, feedback equalization optimization is carried out, and a deep audio equalization model is obtained. Accurate equalizer parameters are generated, and the automatic equalizer parameter generation process is completed. According to the invention, a more efficient equalizer parameter automatic generation method is realized.

The invention provides an audio data processing method, electronic equipment, a storage medium and a computer program product, and the method comprises the steps: extracting audio feature information from target audio data according to a feature extraction strategy corresponding to a specified classification algorithm; classifying the audio feature information according to the specified classification algorithm to obtain a music type of the target audio data; calling a sound effect enhancement strategy matched with the music type, and performing sound effect enhancement processing on the target audio data; wherein the sound effect enhancement strategy refers to an adjustment mode of an equalizer, a reverberation mode and a bass enhancement function. According to the scheme, classification and sound effect enhancement of the audio data are realized under the condition of relatively low hardware requirements and resource consumption.

A method for adapting a continuous time equalizer (CTE) includes determining a gain of a discrete time equalizer (DTE) and determining whether the gain has increased or decreased by more than the threshold amount. Responsive to determining that the gain has increased or decreased by more than the threshold amount, the method includes sequentially configuring the CTE for multiple CTE settings such that gain of the CTE is caused to increase or decrease in a same direction with the change in gain of the DTE. The method also includes determining a separate figure of merit (FOM) for each of the multiple CTE settings and selecting a new CTE setting from the multiple CTE settings based on the FOM for each of the multiple CTE settings.

A semiconductor device is provided. The semiconductor device includes: an equalizer circuit configured to output a first control signal corresponding to a first bit of original two-bit data and a second control signal corresponding to a second bit of the original two-bit data; and a driver circuit including a plurality of pull-up transistors connected between an output node and a first power node configured to provide a first power supply voltage, and a plurality of pull-down transistors connected between the output node and a second power node configured to provide a second power supply voltage, wherein the second power supply voltage is lower than the first power supply voltage, and the driver circuit is connected to the equalizer circuit in series. The plurality of pull-up transistors includes a first pull-up transistor and a second pull-up transistor connected to each other in parallel, between the first power node and the output node, and a third pull-up transistor and a fourth pull-up transistor connected to each other in series, between the first power node and the output node.

The invention relates to the field of underwater acoustic communication, in particular to an AFDM underwater acoustic communication channel equalization method based on adaptive decision feedback, aiming at an underwater acoustic channel with long time delay, large Doppler and remarkable time-varying property, an adaptive decision feedback equalizer can timely track the channel change and improve the symbol detection performance by updating the equalizer weight coefficient; the performance of a communication system is improved, error propagation can be reduced, and more reliable underwater information transmission can be realized. As the adaptive decision feedback equalizer belongs to a symbol-level equalization method, the complexity of the adaptive decision feedback equalizer is linearly related to a modulation symbol number N and the size of a filter. Therefore, the calculation complexity of the method is lower than that of other equalization algorithms, and the low calculation complexity better meets the underwater communication requirement with limited calculation resources. According to the method, the transmission performance of underwater acoustic communication is improved, and good balance can be achieved between low complexity and good performance.

The invention relates to the technical field of optical fiber communication, and discloses a clock recovery system and method for a multi-aperture coherent optical signal, and the system comprises an interpolator which obtains coherent optical signals received through a plurality of different apertures, and carries out the preliminary adjustment of the clock phase of the coherent optical signals; the MIMO equalizer equalizes and synthesizes a plurality of coherent optical signals of which the clock phases are readjusted, and the error detector tracks the clock phases of the transmitted coherent optical signals in real time and extracts the clock phase error of each coherent optical signal relative to the transmitted coherent optical signals in the synthesized coherent optical signals. The clock generation chip is used for generating a reference clock with a corresponding clock phase according to the clock phase error of each coherent optical signal, so that the clock phase of each coherent optical signal is readjusted through the reference clock, and at the moment, the coherent optical signals with the readjusted clock phases are balanced and synthesized; the quality of the synthesized multi-aperture signal can be improved.

The invention relates to a parameter determination method and device of a microstrip gain equalizer and computer equipment, and relates to the technical field of microwave passive circuits. The method comprises the following steps: obtaining first fitness corresponding to each initial equalizer parameter configuration scheme according to each first equalizer configuration parameter contained in the initial equalizer parameter configuration scheme; based on the first fitness corresponding to each initial equalizer parameter configuration scheme, updating the initial equalizer parameter configuration scheme to obtain a target equalizer parameter configuration scheme; obtaining second fitness corresponding to each target equalizer parameter configuration scheme according to each second equalizer configuration parameter contained in each target equalizer parameter configuration scheme; and taking each second equalizer configuration parameter contained in the target equalizer parameter configuration scheme corresponding to the minimum second fitness as the equalizer configuration parameter of the microstrip gain equalizer. By adopting the method, the parameter determination efficiency in the design process of the microstrip gain equalizer can be improved.

The invention is suitable for the technical field of audio processing, and provides a sound effect adjusting method and device, intelligent equipment, a storage medium and a program product, and the method comprises the steps: receiving a sound effect adjusting request which comprises an identifier of to-be-adjusted audio data and sound effect demand description information of to-be-adjusted audio; the sound effect demand description information is used for indicating the sound effect expected by the user; in response to the sound effect adjustment request, determining a target audio parameter of a sound effect equalizer of the intelligent device based on the sound effect demand description information; according to the target audio parameter, a sound effect equalizer is controlled to carry out sound effect optimization on the to-be-adjusted audio data, target sound effect data is obtained, and the target sound effect data conforms to the sound effect expected by the user; and controlling an audio playing device of the intelligent equipment to play the target sound effect data. According to the scheme, the sound effect adjustment time is shortened, the situation of improper parameter setting is also reduced, the intelligence and convenience of sound effect adjustment are improved, and the sound effect adjustment efficiency and the sound effect adjustment effect are improved.

The invention provides a parameter tuning method, device and system for an equalizer, and the method comprises the steps: carrying out the downsampling of a first parameter, obtaining the first downsampling data of the first parameter, determining a first generation tuning value of the first parameter according to the first downsampling data and the first quality information in an electric signal in a PHY, and carrying out the tuning of the first generation tuning value of the first parameter; the first-generation tuning value is used for configuring the equalizer; then, first candidate data of the first parameter are obtained according to the first-generation tuning value of the first parameter and the first deviation range of the first parameter, a second-generation tuning value of the first parameter is determined according to the first candidate data and second quality information in the electric signal in the PHY, and the second-generation tuning value is used for updating the first-generation tuning value to configure the equalizer. Therefore, the tuning result of the joint tuning of all the to-be-configured equalization parameters can be determined by scanning fewer parameters for a small number of times and taking the quality of the corresponding electric signal as a basis, and rapid and convenient joint tuning of the equalization parameters is realized.

The invention provides an audio parameter setting method and an electronic device. The method comprises the following steps: acquiring a sound signal played by a loudspeaker device; a frequency response curve of the sound signal is generated. And smoothing the frequency response curve of the sound signal to obtain a smoothed response curve of the sound signal. Determining a target frequency response curve according to the sound signal; and detecting a plurality of turning points of the smoothed response curve of the sound signal by using the neural network model. And determining an equalizer parameter according to a plurality of turning points of the smoothed response curve and the target frequency response curve. Therefore, the audio quality can be improved.

A detection method includes: obtaining a decision feedback equalizer coefficient, where the decision feedback equalizer coefficient includes a tap coefficient; obtaining a decision signal sequence of a decision feedback equalizer; determining a first location of a decision signal of a start of burst error in the decision signal sequence when the tap coefficient is less than or equal to a first preset threshold; and determining a second location of a decision signal of an end of burst error in the decision signal sequence based on the first location.

The embodiment of the invention provides a channel training method and a program product, and the method comprises the steps: responding to a first updating instruction and first equalizer state information sent by first equipment, and carrying out the receiving end adaptive equalization of an equalization coefficient through a software interface; sending a second updating instruction and second equalizer state information to the first equipment according to the self-adaptive equalization result of the receiving end, so that the first equipment modifies a specified equalization coefficient, and receiving third equalizer state information and a third updating instruction sent after the first equipment is updated; and in response to the third equalizer state information and the third update instruction, hardware adaptation is performed on the equalization coefficient of the receiving end, fourth equalizer state information is sent to the first device according to a hardware adaptation result, and the fourth equalizer state information is used for indicating a channel formed by the first device and the second device to complete training. The problems of unstable channel training and long time consumption caused by channel training only through hardware automatic training in related technologies are solved.

The invention provides a constant modulus blind equalization method based on chaotic mapping and Gaussian perturbation, which comprises the following steps of: 1, estimating a receiving signal of a receiving end of a channel simulator to obtain a constant value; initializing related parameters; 2, entering a particle swarm optimization (PSO) main cycle; step 3, according to the chaos Sine mapping and the upper and lower limits of the inertia weight, obtaining the inertia weight and the learning factor of the current iteration number; 4, calculating fitness values of the particles; 5, updating the speed and the position of the particle; step 6, repeating the steps 2-5 until a termination condition is met; and step 7, obtaining a global optimal fitness value and a global optimal position after a termination condition is satisfied, and calculating an output signal according to an equalizer constructed at the global optimal position at the moment. According to the method, a logarithmic function is introduced to construct a nonlinear asynchronous learning factor which is different from a constant learning factor in an original algorithm, so that the global development capability and the local search capability of the algorithm are balanced.

The invention relates to an impedance determination method and device, computer equipment, a readable storage medium and a program product. The method comprises the steps that first eye diagrams of an on-chip terminal are acquired, and the first eye diagrams are eye diagrams of an electric signal under target equalizer parameter conditions corresponding to candidate impedance values of the on-chip terminal; each candidate impedance value corresponds to a target equalizer parameter; according to a first preset condition and the eye diagram parameters in the first eye diagrams, obtaining a target eye diagram from the first eye diagrams; wherein the first preset condition represents a screening condition related to the eye pattern parameter; and taking the candidate impedance value corresponding to the target eye diagram as a target impedance value. According to the method, the target impedance value is adopted to design the chip system, so that the performance of the chip is improved.

A system for activating multi-rate optical network units (ONUs) includes an optical network having an optical line terminal (OLT) optically coupled to a plurality of ONUs, including one or more multi-rate ONUs configured to support multiple upstream rates. During discovery when other ONUs are permitted to transmit in the same quiet window, a multi-rate ONU that supports a higher rate is instead configured to send a discovery response at a sufficiently low rate so as not to require training of an equalizer, thereby allowing for a short preamble for the discovery response. Thereafter, during a quiet window for ranging when only the multi-rate ONU will be transmitting to prevent data collisions, the multi-rate ONU is transitioned to a higher rate and transmits a message with a sufficiently long preamble to train an equalizer at the OLT.

The invention provides a method and a system for establishing an LED (light-emitting diode) model represented by driving current and temperature, and belongs to the technical field of visible light communication. An equivalent circuit model of the LED and analytic solutions of all parts of the model about the driving current and the temperature are obtained from the small signal form and the steady state form of the rate equation respectively; a model related to the carrier experience time is further provided, so that the analytical solution of the model only comprises the driving current and the temperature, the model coefficient is few, and the undetermined coefficient can be solved only through three actual measurement results at least; impedance and frequency response predicted by the model have high conformity with an actual measurement result, and meanwhile, a system can be guided to compensate nonlinearity of the LED according to the LED working state predicted by the model, for example, a dynamic equalizer is designed to enable poles of the equalizer to be matched with the LED under the current working condition at any time, so that the optimal equilibrium state is achieved.

A signal processing device acquires information relating to a stringed instrument, and sets a parameter to an equalizer based on the acquired information relating to the stringed instrument.

The invention relates to a dynamic self-adaptive communication, sensing and calculation integrated method for an air-space-ground integrated network. The method comprises the following steps: S1, constructing an air-space-ground integrated network model fusing a low earth orbit satellite, an unmanned aerial vehicle and ground users / edge nodes; s2, establishing a cross-layer communication-perception-calculation integrated joint optimization model, and respectively forming two modes of binary unloading and partial unloading according to whether a task can be segmented or not; s3, for a binary unloading mode, realizing distributed solution and generating an unloading decision and a beam / power / bandwidth / computing resource scheme by using a dynamic adaptive prediction alternating direction multiplier (DAP-ADMM); and S4, aiming at a partial unloading mode, adopting a weighted minimum mean square error alternative optimization method, introducing a receiving equalizer, error weight and other auxiliary variables, executing block coordinate descent iteration, and outputting an unloading proportion, power distribution and a beam forming result. According to the DA-ISCC architecture and the optimization algorithm thereof provided by the invention, cross-domain calculation unloading with low time delay, high throughput and low overhead can be realized in a high-dynamic space-air-ground integrated network environment, and the DA-ISCC architecture and the optimization algorithm thereof can be widely applied to emergency communication, low-altitude logistics, ocean monitoring and other scenes.

The present invention provides a method and system for establishing an LED model characterized by a driving current and a temperature, belonging to the technical field of visible light communication. The method starts from a rate equation of LED carrier motion, obtains an equivalent circuit model of the LED and analytical solutions of each part of the model with respect to the driving current and the temperature from the small signal form and the steady-state form of the rate equation respectively; and further proposes a model of carrier experience time so that the analytical solution of the model only includes the driving current and the temperature, and the model coefficients are few, and the coefficients to be determined can be solved with at least three actual measurement results; the impedance and frequency response predicted by the model have a high degree of conformity with the actual measurement results, and at the same time, the system can be guided to compensate for the nonlinearity of the LED based on the LED working state predicted by the model, such as designing a dynamic equalizer so that the equalizer poles can be matched with the LED of the current working conditions at any time, so as to achieve the best equilibrium state.

A method, system, device and medium for optimizing the design of an analog equalizer based on a deep learning algorithm. The method first trains an inverse neural network using the electromagnetic response curve of the equalizer circuit. Additional equalizer circuit parameter data is generated using the trained inverse neural network and combined with the original simulation data into a new data set to solve the problem of excessive simulation software time consumption. A forward neural network is trained using this new data set. The trained forward neural network outputs a predicted equalizer electromagnetic response curve, which is compared with a target equalizer electromagnetic response curve. If conditions are met, the inverse neural network is used to generate actual equalizer circuit parameters that meet the conditions, and the design is completed. The system, device and medium are used to carry and implement the method. The present invention optimizes the design process through a deep learning method, realizes the automation of circuit design, and reduces design time and computational complexity.

This application provides a location detection method and a related apparatus, and relates to the field of communication technologies. The method includes: obtaining a decision feedback equalizer coefficient, where the decision feedback equalizer coefficient includes a tap coefficient; obtaining a decision signal sequence of a decision feedback equalizer; determining a first location of a decision signal of a start of burst error in the decision signal sequence when the tap coefficient is less than or equal to a first preset threshold; and determining a second location of a decision signal of an end of burst error in the decision signal sequence based on the first location. In this application, the corresponding location detection method is determined based on a relationship between the tap coefficient and a preset threshold, and a location of a decision signal corresponding to a burst error in the decision signal sequence is accurately determined according to the location detection method, to help eliminate the burst error in the decision signal sequence.

A method of decision feedback equalization includes receiving symbols at a first rate, converting the symbols into first bits and second bits using a first slicer and a second slicer, respectively, wherein each of the first and second slicers operates at a second rate approximately equal to half the first rate, alternately sampling the first bits using a first latch and a second latch, wherein each of the first and second latches operates at a third rate approximately equal to a quarter of the first rate, alternately sampling the second bits using a third latch and a fourth latch, wherein each of the third and fourth latches operates at the third rate, alternately coupling the first latch and the second latch to a feedback input of the second slicer, and alternately coupling the third latch and the fourth latch to a feedback input of the first slicer.

The invention provides an uplink MIMO-NOMA (Multiple Input Multiple Output-Non-Orthogonal Multiple Access) system optimization method with a maximized sum rate, which comprises the following steps: firstly, in order to suppress interference between a transmission cluster and a reflection cluster, providing an inter-cluster SIC method: clustering users according to transmission characteristics, adopting space division multiple access in each cluster, and performing inter-cluster SIC of NOMA transmission so as to reduce the interference between the clusters; secondly, a non-convex optimization problem aiming at maximizing the total rate of the system is formulated and solved by using an alternating optimization framework, in the framework, a receiving equalizer is optimized according to a minimum mean square error standard, and transmitting power control of a user is processed through successive convex approximation, so that sub-problems are easy to process; and finally, the beam forming of the active STAR-RIS adopts a sequential rotation design. According to the invention, the dual-fading effect is effectively overcome by adjusting the phase shift and amplitude of electromagnetic waves, an alternating optimization framework is also provided to solve the described non-convex problem, user power is controlled through continuous convex approximation, and active STAR-RIS beam forming is realized through sequential rotation.

The invention discloses a decision feedback equalizer and a semiconductor device. The semiconductor device includes: a data pad; and a decision feedback equalizer configured to equalize the data signal received through the data pad. The decision feedback equalizer may include a plurality of decision feedback equalization blocks respectively corresponding to a plurality of clock signals having different phases. Each of the plurality of decision feedback equalization blocks may include: an input stage configured to compare a reference voltage with a data signal based on a corresponding clock signal, amplify a difference between the reference voltage and a voltage of the data signal, and output the amplified difference; and generate a first feedback signal and a second feedback signal based on the amplified difference between the reference voltage and the voltage of the data signal; and a latch stage configured to generate a first output signal and a second output signal based on a difference between the first feedback signal and the second feedback signal.

The invention provides a PAM4 signal equalizer based on automatic gain control, which comprises a continuous time linear equalizer CTLE, a variable gain amplifier VGA, a negative capacitor I, a negative capacitor II, a power detector, an encoder, an eye diagram parameter calculation module, a digital time converter, a pipelined analog-to-digital converter and a counter, the first input end of the continuous time linear equalizer CTLE is connected to an input signal DIN, the second input end VCTLE1 of the continuous time linear equalizer CTLE is connected to the first output end of the encoder, the third input end VCTLE2 of the continuous time linear equalizer CTLE is connected to the second output end of the encoder, and the output end N1 of the continuous time linear equalizer CTLE is connected to the input end of the first negative capacitor and the first input end of the VGA. According to the invention, optimization and real-time adaptive adjustment for PAM4 characteristics can be realized, the adaptability and the anti-noise capability can be improved, and the signal recovery capability and the signal integrity can be improved.

The invention discloses a coding method for reducing power consumption of a feed-forward equalizer, and relates to the technical field of equalizer coding, and the coding method is characterized in that each bit is only a value {-1, 0, + 1}; adjacent bits cannot be non-zero at the same time; a first non-zero bit symbol from a high bit to a low bit must be consistent with the overall symbol; the code decomposes the input data into a few symbolized weight bits, and multiplication of the input data and the weight can be converted into shift and addition or subtraction operation. Therefore, for a single tap, corresponding multiplication operation can be realized through shifting and addition or subtraction, and final output can be obtained through summation finally. According to the invention, equivalent multiplication can be completed through a limited number of shift and addition; during implementation, a multiplier is not needed, only symbol control addition is needed, and a very efficient digital system is achieved through the encoding mode and is used for achieving multiplication, especially on occasions needing high throughput rate and low delay.

A device according to the present invention includes: one or more calculation units configured to calculate correlation sums of binary divided first and second parts of the search range specified for a series of discrete digital signals having a unit time difference from each other, the discrete digital signals being specified based on a set code constant sequence, and a controller configured to specify as the search range one part in which a dominant value appears in one or more correlation sums for the first part and one or more correlation sums for the second part, each of which is calculated by the one or more calculation units, and cause the one or more calculation units to recalculate the correlation sums for each of the binary divided sub parts of the one part.

Embodiments described herein relate to a plenum including an inlet configured to receive fluid from a coolant system, a plurality of outlets configured to be fluidly coupled to a plurality of heat exchangers, and a tapered portion corresponding to the plurality of outlets, the tapered portion configured to maintain the fluid at a desired velocity. In some embodiments, the tapered portion may include a first portion defining a first tapering rate and a second portion defining a second tapering rate. In some embodiments, the second tapering rate may be lower than the first tapering rate. In some embodiments, the inlet may include two inlet ports.