15 results about "Dsp algorithms" patented technology

The invention relates to the technical field of intelligent adjustment of audio equipment, and discloses a system for automatically adjusting loudspeaker volume and sound effect according to wind sound volume, which comprises a wind noise acquisition module, an audio processing module, a main control module and a loudspeaker module, and is characterized in that the modules are sequentially in signal transmission connection. The loudspeaker volume and the sound effect type are dynamically adjusted according to the wind noise level, the wind noise is not counteracted to retain environment sound perception, and the problems that an existing noise reduction scheme is not real, influences safety and is not adaptive to the wind noise adjustment sound effect are solved. According to the invention, the wind noise acquisition module adopts a loudspeaker periphery distributed microphone layout, and captures a wind noise and environmental noise mixed signal near a loudspeaker at 360 degrees, thereby avoiding irrelevant noise interference; the audio processing module separates and recognizes the wind noise through a special DSP algorithm, and accurately judges the wind noise level by combining with the comparison of a wind noise feature database, thereby providing a reliable basis for adjustment, and overcoming the defects of inaccurate collection and fuzzy recognition of traditional equipment.

This invention relates to a method for modeling audio response curves in a web environment, comprising the following steps: S1: Generating frequency and gain based on screen click events, and forming control points by user input of Q values; or obtaining three parameters—frequency, gain, and Q value—input by the user to form control points; S2: Converting the frequency, gain, and Q value of the control points into parameters of a Gaussian function and generating connection points, smoothing the connection points with a Bézier curve, and then drawing the audio response curve; S3: Mapping the Gaussian function parameters to a Web Audio API and obtaining a digital audio signal through a DSP algorithm, and then converting it into an analog audio signal. By simulating the peak value and gain characteristics of the audio response curve in local regions using a Gaussian function, a natural transition of the audio response curve in local regions is ensured; by using a Bézier curve to smoothly connect the Gaussian curves, a highly accurate and sufficiently smooth audio response curve is obtained.

The present invention provides a simultaneous and co-frequency millimeter wave communication perception fusion system and method. The system includes a CU, a DU, n RUs, and m users corresponding to each RU. The communication perception fusion method first utilizes a photonics-assisted network architecture to up-convert perception and communication waveforms to millimeter wave frequencies. At the user end, a corresponding DSP algorithm eliminates interference from radar signals on communication signals and extracts communication signals from the communication perception waveform. At the radar receiving end, a de-chirping mechanism is used to extract perception information from the millimeter wave communication perception waveform. The radar and communication waveforms used are consistent with existing independent synaesthesia systems, requiring no specific modifications and easily compatible with existing synaesthesia algorithms. The simultaneous and co-frequency synaesthesia and perception fusion of millimeter wave signals improves system spectrum efficiency and avoids perception or communication interruptions. At the communication receiving end, a secondary de-skewing DSP algorithm is used to eliminate the impact of chirping in the perception waveform on communication.

The invention relates to an audio response curve modeling method based on a Web environment, and the method comprises the following steps: S1, generating frequency and gain according to a screen click event, and inputting a Q value by a user to form a control point; or acquiring three parameters of frequency, gain and Q value input by a user to form a control point; s2, converting the frequency, the gain and the Q value of the control point into parameters of a Gaussian function, generating a connection point, smoothing the connection point through a Bezier curve, and then drawing an audio response curve; and S3, mapping the Gaussian function parameter to a Web Audio API, obtaining a digital audio signal through a DSP algorithm, and converting the digital audio signal into an analog audio signal. The peak value and gain characteristics of the audio response curve in the local area are simulated through a Gaussian function, and natural transition of the audio response curve in the local area is ensured; and performing smooth connection on the Gaussian curves by using a Bezier curve to obtain an audio response curve which is high in accuracy and is smooth enough.

The application relates to the field of Beidou positioning technology, in particular to a Beidou orientation system and method, which comprises a central processing module, two Beidou baseband modules and two Beidou antennas arranged on a Beidou orientation device; the two Beidou baseband modules are connected with the two Beidou antennas in one-to-one correspondence respectively; the central processing module comprises an ARM service processing unit and a DSP algorithm processing unit; a communication connection is established between the ARM service processing unit and the DSP algorithm processing unit; the ARM service processing unit is connected with the two Beidou baseband modules respectively; by using the mode of an ARM core and a DSP core, the two core processing data are relatively independent, and synchronous concurrent processing of data receiving and sending and data algorithm can be realized; data interaction is carried out between the ARM service processing unit and the DSP algorithm processing unit through a shared memory mode, and efficient data processing can be realized.

This invention relates to the field of power system parameter monitoring technology and discloses a high-precision analysis method for electrical parameters based on a lightweight DSP algorithm, comprising the following steps: sampling and data buffering, lightweight DSP core processing, electrical parameter output, and code-level optimization. This invention suppresses spectral leakage through integer-cycle sampling combined with window functions, achieving harmonic measurement accuracy of ±0.5%. After Kalman filtering, data fluctuation is reduced by 80%, and steady-state accuracy is improved to within 0.1%, enabling high-precision measurement. Simultaneously, downsampling and integer operations reduce computational load, and code-level optimization keeps CPU utilization ≤15%, allowing it to run on low-cost MCUs such as the STM32F407.

The present invention discloses an active enhancement detection method for line fault characteristics of a distribution box, which relates to the technical field of power system detection, and includes the following steps: Collect electrical signals in the distribution box in real time, and process the signals through an active enhancement strategy to highlight fault characteristics, and at the same time obtain the enhanced electrical signal data. The present invention ensures that the enhanced signal spectrum does not overlap with the system operating frequency through real-time signal acquisition, enhancement, feature extraction, frequency overlap analysis, machine learning evaluation, and digital signal processing (DSP) frequency shift operation, avoiding signal distortion or misjudgment of fault signals caused by spectrum interference. Combining a machine learning model with a DSP algorithm, dynamically adjust signal processing, automatically identify and correct frequency overlap problems, improve the performance and stability of the fault detection system, provide an intelligent and accurate fault detection solution, and enhance the safety and reliability of the power system.

The invention relates to the technical field of Beidou positioning, in particular to a high-precision Beidou orientation method, which comprises a central processing module arranged on Beidou orientation equipment, two Beidou baseband modules and two Beidou antennas, and the two Beidou baseband modules are respectively connected with the two Beidou antennas in a one-to-one correspondence manner. The central processing module comprises an ARM service processing unit and a DSP algorithm processing unit, communication connection is established between the ARM service processing unit and the DSP algorithm processing unit, the ARM service processing unit is respectively connected with the two Beidou baseband modules, data processing of the two cores is relatively independent by using a mode of an ARM core and a DSP core, and the two cores are respectively connected with the two Beidou baseband modules. Synchronous concurrent processing of data receiving and transmitting and data algorithms can be really realized; data interaction is carried out between the ARM service processing unit and the DSP algorithm processing unit in a memory sharing mode, and efficient data processing can be achieved.

A fitting device for a hearing aid earphone comprises an earphone main body and a remote controller, the earphone main body and the remote controller are connected through signals, and the earphone main body is internally provided with an MCU control unit, a DSP algorithm processor, a memory, a signal amplification processor, a microphone array gain regulator, an analog output gain regulator and a power supply module. Wherein the MCU control unit is in signal connection with a remote controller, is in output connection with the signal amplification processor, the microphone array gain regulator and the analog output gain regulator, and receives signal input of the memory and the DSP algorithm processor, and the power supply module supplies power to the earphone main body; the remote controller comprises a built-in MCU controller, a pure tone generation module, an LCD display screen and keys, the LCD display screen and the keys are arranged on a shell of the remote controller, the keys are connected to the input of the MCU controller, and signals of the MCU controller are output to the pure tone generation module and the LCD display screen.

The invention provides an optical splitter port identification method and system based on CD estimation in a coherent PON, and relates to the technical field of optical fiber communication. Optical splitter ports are distinguished by cascading devices with different negative CD values at different ports of an optical splitter in an ODN. And a receiving end extracts and estimates the port CD through a two-stage dispersion estimation algorithm, so that identification of different ports is realized, and construction of the network topology is completed. According to the invention, under the condition that the receiving and transmitting modules are not added on the receiving and transmitting sides, the digital signal processing DSP algorithm of the receiving end in the optical network unit ONU is utilized to realize the accurate identification of the optical splitter port.

The invention relates to a direct current zero leakage protector based on artificial intelligence, compared with a conventional alternating current zero leakage protector, the zero leakage protector expands the alternating current range to the direct current field, and a traditional relay element and a mechanical release are abandoned in a protection device; in the control mode, a high-speed microprocessor and an MOS tube based on the AI technology are adopted, fixed-point, floating-point and DSP algorithms and AI reasoning operation are supported, and the action response time is at least one order of magnitude faster than that of a traditional zero leakage protector. In a topological structure, a control box and a junction box are combined into a whole, an input digital current transformer adopts a horizontal form, an output digital current transformer adopts a vertical form, and cross induction interference between electromagnetic fields is overcome to the greatest extent; the invention is suitable for both an alternating current circuit and a direct current circuit, is particularly safe, and can be widely applied to various fixed or movable high-power electrical equipment.

The invention relates to the technical field of wireless communication, and provides a terahertz wireless communication system based on dual-polarization multiplexing and MIMO, which comprises a photon-assisted dual-polarization terahertz signal generator, a dual-polarization multiplexing MIMO frame and a structure receiving end digital signal processing module, and is characterized in that the system is combined with a dual-polarization multiplexing MIMO frame to improve the signal directivity and the transmission distance; and a receiving end adopts a total blind DSP algorithm to realize signal recovery without a pilot frequency sequence, and weights are allocated through an MIMO-MRC technology to merge two paths of signals. The system is suitable for 6G ultra-high-speed wireless access, space-air-ground integrated network and satellite communication scenes, optical fiber and wireless transmission advantages are fused through a photon-assisted technology, coherent destruction caused by a multipath effect is inhibited, polarization sensitivity is reduced, communication capacity and transmission distance are remarkably improved, and a reliable technical path is provided for Tbps-level communication in a terahertz frequency band.

The invention discloses a line fault characteristic active enhancement detection method for a distribution box, and relates to the technical field of power system detection, and the method comprises the following steps: carrying out the real-time collection of an electrical signal in the distribution box, carrying out the processing of the signal through an active enhancement strategy, highlighting the fault characteristic, and obtaining the data of the enhanced electrical signal. Through real-time signal acquisition, enhancement, feature extraction, frequency overlap analysis, machine learning evaluation and digital signal processing (DSP) frequency shift operation, it is ensured that an enhanced signal frequency spectrum is not overlapped with a system working frequency, and fault signal distortion or misjudgment caused by frequency spectrum interference is avoided. A machine learning model and a DSP algorithm are combined, signal processing is dynamically adjusted, the frequency overlapping problem is automatically identified and corrected, the performance and stability of a fault detection system are improved, an intelligent and accurate fault detection solution is provided, and the safety and reliability of a power system are enhanced.

The invention relates to the technical field of satellite navigation, in particular to an ionospheric scintillation signal simulation monitoring system and method in Beidou / GPS, after initial information is issued by an upper computer, a baseband signal board processes the initial information to generate a GPS / BDS simulation intermediate frequency signal; a Cornell model is built in a DSP algorithm of a baseband signal board, an ionosphere delay model and a troposphere delay model are built at the same time, and finally ionosphere scintillation signal simulation is carried out. A GPS / BDS intermediate frequency signal is received through a signal processing module; and the upper computer software module performs man-machine interaction, defines the working environment of the user receiver and displays and controls data of the GPS / BDS simulator. The upper computer software module provides a good man-machine interaction interface, so that a user can conveniently define the working environment of a user receiver and display and control data of a GPS / BDS simulator. Therefore, the problem that the measurement precision is affected by signal propagation delay caused by ionospheric flicker is solved.

The invention discloses a modularly designed mixed signal acquisition and processing system and method, and belongs to the technical field of mixed signal acquisition, the system comprises a core board, a bottom board and a pluggable module, the bottom board is provided with a programmable clock unit and a sampling trigger and data channel, the core board is provided with a DSP algorithm module, a storage and calibration management module and a peripheral output interface. The programmable clock unit is used for generating a reference clock, the sampling trigger and data channel is used for receiving a signal input by the pluggable module, and the DSP algorithm module is used for performing time domain and frequency domain processing on sampled data; the storage and calibration management module is used for storing configuration parameters, calibration data and operation logs, the pluggable module is used for realizing input conditioning or output conversion of signals, and the peripheral output interface is used for providing a control and synchronization interface for an external device. According to the invention, the signal processing precision and the real-time processing capability can be improved, the system cost is reduced, the maintenance convenience is improved, and the applicability is realized.