80results about How to "Improve intelligibility" patented technology

The invention relates to the technical field of medical data management, in particular to a medical data document classification and marking system which comprises a document structure analysis module, a semantic information identification module, a classification and marking module and a document intelligent label module. According to the method, document visualization and retrieval efficiency is improved by analyzing document content and extracting titles, texts and appendixes, analytic performance and operability of data are enhanced by identifying key information and semantic relations and accurately extracting and labeling medical terms, medicine research data and medical record records, storage is optimized according to priorities in combination with classification and marking, and the method is high in practicability and easy to popularize and use. Clinical values and application scenes of documents are clarified, document management processes are refined, management compatible with clinical data formats is formed through structured storage of marked data and classification label generation, storage normalization and retrieval efficiency are improved, the requirement for accurate information calling in clinical tests is met, document management capacity is improved, and the document management efficiency is improved. And support is provided for improvement of clinical decision-making and research efficiency.

The invention discloses an electric power knowledge question-answering system construction method and system combined with engineering ethical analysis, and the construction method comprises the steps: constructing all modules, and carrying out the fine adjustment of a large language model based on a P-Tuning method, so as to construct an electric power system question-answering knowledge base; the question understanding and classification step is responsible for receiving input questions of a user and performing semantic analysis, technical classification and ethical risk identification on the input questions; searching background knowledge; carrying out engineering ethical analysis; generating questions and answers; and a feedback and improvement step: entering a model tuning stage, and carrying out adaptive tuning on the model by adopting a reinforcement learning framework. According to the method, comprehensive engineering ethical analysis can be carried out on technical schemes and operation suggestions, finally, answers generated by the system are accurate and feasible in technology and have ethical compliance at the same time, it is ensured that a user can balance technical requirements and ethical responsibility in practical application, the intelligent level of an electric power knowledge question-answering system is remarkably improved, and the user experience is improved. And important support is provided for safe operation and sustainable development of the power industry.

The invention discloses a fire-fighting long-distance ad hoc network full-duplex voice communication system, which relates to the technical field of voice communication, and realizes whole-staff real-time voice acquisition through cooperative work of a microphone array module, a noise sensing module, a noise filtering module, a communication state evaluation module and a signal gain adjustment module. Two-way voice communication between firefighters and between the firefighters and a command center is ensured, in a fire rescue site, the noise sensing module can accurately recognize various background noise in a fire environment, such as explosive sound or equipment operation sound, and noise interference areas are classified and marked in time. The noise filtering module can dynamically adjust the noise suppression coefficient according to the noise type and intensity through a self-adaptive noise suppression algorithm, interference signals are effectively removed, and through real-time and dynamic noise suppression, firemen can clearly receive command information in a complex environment, and misunderstanding or communication faults caused by noise interference are reduced.

The invention discloses a voice noise reduction system in a multi-noise environment, and particularly relates to the technical field of voice noise reduction, which comprises a basic division module for dividing voice data into a plurality of basic voice segments with the same duration; the rationality analysis module judges whether segmentation points of adjacent fragments are reasonable or not, and if the segmentation points are not reasonable, the optimization marking module searches for new segmentation points for optimization; the loop module repeatedly verifies the optimization result until all the voice segments are reasonably segmented; the noise reduction module extracts voice and noise features, and performs noise reduction processing by using a noise reduction strategy selection model; the recombination module recombines the voice segments after noise reduction into complete voice output; according to the invention, the voice signal is divided into a plurality of segments, the most suitable noise reduction algorithm can be matched for each segment according to the noise feature of each segment, and the method can cope with a more complex and unstable noise environment through a dynamically adaptive noise reduction strategy, and guarantees that the voice quality is better improved in various scenes.

The invention provides a sound processing method, system and device based on a hearing aid and a storage medium, and the method comprises the steps: collecting sound signals, including background noise and voice signals, of a hearing aid use environment in a large scale; the sound signals are preprocessed, abnormal values and missing values are processed, the abnormal values and the missing values are scaled into a unified scale, and the sound signals are divided into a training set and a test set; adjusting and optimizing parameters of the standard adaptive filtering model by using an improved gradient descent optimization algorithm, wherein the parameters comprise a filter order, a maximum number of iterations, an initial step length and a regularization coefficient; and inputting the test set into the optimized algorithm to realize optimization processing of the sound signal. According to the invention, the voice definition and the background noise suppression capability can be improved, and the use effect and the user experience of the hearing aid in a complex environment are improved.

The invention discloses an underground space intelligent layout method and system, and the method comprises the following steps: S1, generating an initial layout through employing a graph theory method based on preset underground space layout knowledge and constraint conditions inputted by a user; s2, optimizing the initial layout through a comprehensive evaluation index by using a multi-objective optimization algorithm; and S3, visually displaying the layout result and the evaluation data. The layout system comprises a layout module, an optimization module and a visualization module, and the optimization module uses a multi-objective optimization algorithm to optimize the generated initial layout through comprehensive evaluation indexes. According to the method, the efficiency of underground space layout design is greatly improved, multiple key indexes such as function compatibility, people flow organization efficiency, average accessibility and space utilization rate are comprehensively considered, and it is ensured that the generated layout scheme achieves optimal balance in all dimensions.

The invention belongs to the field of artificial intelligence and financial science and technology, and relates to a question and answer reply generation method which comprises the steps that a query text is input into a reply generation model, and the reply generation model comprises an input analysis network, a style recognition network, a retrieval enhancement generation network and a reply selection network; processing the query text through the input analysis network to obtain a text input vector; inputting the user historical dialogue data and the text input vector into a style recognition network to generate a style recognition result; inputting the text input vector and the style recognition result into a retrieval enhancement generation network, and generating a plurality of candidate replies in combination with related document fragments retrieved by a preset knowledge base; a final reply is selected from the plurality of candidate replies through the reply selection network. The invention further provides a question and answer reply generation device, computer equipment and a storage medium. In addition, the invention also relates to a block chain technology, and the final reply can be stored in the block chain. According to the method, the content meeting the long-term preference and personalized requirements of the user can be generated.

The invention provides a data monitoring early warning method and system, computer equipment and a storage medium, and the method comprises the steps: obtaining a dotting message, transmitting the dotting message to a gateway controller through a controller domain network protocol, and when the gateway controller recognizes that a received message identifier belongs to a controller domain network message of the dotting message, sending the dotting message to the controller domain network protocol; analyzing the dotting data of the controller domain network message, and calling an application programming interface in a server database to perform dotting on the data of the target electronic control unit; and displaying data of the target electronic control unit on a preset display panel based on the dotting result, and performing monitoring and early warning on the data of the target electronic control unit through index data configured in advance or in real time. By displaying the data of the electronic control unit on the preset display panel, the target personnel can be assisted to monitor and analyze the data of the electronic control unit in real time, so that the manual inspection time is shortened, and the efficiency and quality of the whole vehicle test are greatly improved.

The invention discloses a communication method and device for marine very high frequency equipment, electronic equipment and a storage medium. The method comprises the following steps: determining an original voice signal, performing signal processing on the original voice signal, and determining semantic text data; performing data processing on the semantic text data, and determining effective text data in the semantic text data; generating an effective text signal corresponding to the effective text data, and transmitting the effective text signal to a voice receiving end through a target communication channel; and the voice receiving end determines a target communication voice based on the effective text signal. Based on the technical scheme of the invention, the communication quality of the marine very high frequency equipment can be effectively improved.

The invention relates to the technical field of depth measurement, and discloses a carbonization depth detection method, system and device, and the method comprises the steps: starting drilling equipment with impact and rotation functions, and carrying out the automatic drilling according to preset parameters, the preset parameters comprise the drilling depth, the rotation speed, the impact frequency and the like; when a hole is drilled to a certain depth, a concrete powder sample is automatically collected; then the built-in fan is started, and concrete powder in the drilled hole is automatically blown away; the computer system controls the alcohol phenolphthalein solution to be automatically added into the container, and the color change of the solution is monitored in real time through a built-in camera; the average value of the two drilling depths is calculated to serve as a carbonization depth detection result; and automatically generating a detection report. The system comprises a parameter setting module, a depth acquisition module and a result output module. According to the invention, automation, precision and intelligence of carbonization depth detection are realized; the detection accuracy and efficiency are improved, and the standardization of the detection process and the visualization of data are also realized.

The invention discloses a credit risk assessment and transmission method based on multi-source data and a knowledge map, and the method comprises the steps: collecting credit event data from the multi-source data through an integrated crawler technology and a keyword-based event extraction technology, and carrying out the cleaning and conversion, so as to construct a credit knowledge map. The credit knowledge graph identifies each credit subject through a uniform resource identifier (URI), uses a credit evaluation index system as a data specification, and continuously updates through real-time data acquisition. In addition, the invention further provides a credit risk multi-path transmission composite influence probability calculation model, and credit risk management is optimized by analyzing influence factors of nodes in a credit subject association model.

The invention belongs to the field of interphones, and provides a mobile phone auxiliary type intelligent interphone and an implementation method thereof, and the mobile phone auxiliary type intelligent interphone comprises a mobile phone end assembly which comprises a central processing unit, an audio processing chip, a first connection module, a loudspeaker and a microphone; the interphone end assembly comprises a main control chip, a wireless communication module, a second connection module and a power management module; a radio frequency transceiver chip is arranged in the wireless communication module, data transmission is carried out between the second connection module and the first connection module, and the audio processing chip adopts a CNN noise reduction model to process digital audio signals; according to the method, the digital audio signals are processed through the CNN noise reduction model, various complex background noises can be accurately identified and removed, clear voice signals can be restored, the voice definition and intelligibility are greatly improved, it is ensured that both communication parties can accurately transmit information in a noisy environment, and the user experience is improved. The problem of poor voice quality caused by noise interference in a traditional matching mode is effectively solved.

The invention discloses a Tibetan language Andole speech enhancement method suitable for specific noise and general noise, and relates to the technical field of low-resource language speech processing. The method specifically comprises the following steps: acquiring noisy voice information; preprocessing the noisy voice information, and dividing the processed data into a training set and a test set; a noise spectrum feature input by the noise classification module is optimized based on a weighted average method of a logarithmic power spectrum domain hard threshold; constructing a noise classification task and noise suppression task fused multi-target Tibetan speech enhancement model; training and testing the built speech enhancement model by using the training set and the test set to obtain a trained speech enhancement model; and inputting the actually collected noisy speech into the trained speech enhancement model to obtain an enhanced logarithmic power spectrum. Through effective fusion of the noise classification task and the suppression task, the quality and the intelligibility of the Tibetan language Andor dialect enhancement voice are improved, the human-computer interaction of the Tibetan language Andor dialect is more perfect, and the method has a positive effect on popularization of Tibetan language informatization.

This invention provides a speech synthesis method, apparatus, electronic device, and storage medium. The method includes: acquiring target text; performing prosodic level prediction on the target text to obtain prosodic boundary information of the target text; determining each prosodic word in the target text based on the prosodic word boundaries in the prosodic boundary information; acquiring the dialectal phonology corresponding to the target text, and determining the dialectal tone sandhi pattern corresponding to each prosodic word based on the dialectal phonology, wherein the dialectal tone sandhi pattern is used to indicate the tone sandhi boundary in each prosodic word; and synthesizing the dialectal speech corresponding to the target text based on the prosodic boundary information and the dialectal tone sandhi pattern. The speech synthesis method, apparatus, electronic device, and storage medium provided by this invention can improve the accuracy and intelligibility of speech synthesis.

This invention discloses an individual motion behavior analysis system based on dynamic modal configuration and rule-guided fusion, belonging to the fields of intelligent perception, sports science, and human-computer interaction. It includes a data input module, a dynamic modal organization and configuration module, a multimodal feature modeling and fusion module, an indicator-level evaluation and behavior state prediction module, a statistical rule-guided analysis and result correction module, and a structured result generation and output module. These modules work together to form a complete closed loop for motion behavior analysis technology. This invention aims to address the problems in existing motion behavior analysis technologies, such as limited dimensionality of single-modal information, rigid multimodal fusion methods, insufficient stability and interpretability of analysis results, and strong dependence on large-scale manually labeled data.

The invention belongs to the technical field of tunnel detection, and particularly relates to a laser point cloud color giving method based on a linear array image, which comprises the following steps of: 1, scanning and preprocessing the linear array image, generating a laser intensity image by using point cloud reflection intensity information of a laser section scanner, storing laser intensity of all points of a section in each column of the image, and obtaining a laser intensity image; the columns are distributed in the mileage direction; 2, image mileage direction alignment is carried out, and the linear array image and the section laser intensity image represent data of one section in each column and are arranged along the mileage direction; 3, longitudinally stretching and correcting the image; 4, performing multi-source image feature matching; and 5, constructing a pixel coordinate transformation relation. According to the method, projection consistency of the laser intensity image and the optical image is achieved through re-projection, then accurate registration of the laser intensity image and the optical image is achieved through feature matching, and finally unification of the laser intensity image and the optical image is achieved through a polynomial model, so that corresponding color information is found for each laser point cloud.

The invention discloses a speech enhancement method based on a time sequence convolutional network, and the method comprises the steps: obtaining a data set, completing the preprocessing of the data, and generating a data pair for training a model; constructing a measurement standard of the system according to general industry knowledge rules and actual application scene requirements; constructing an encoder by using convolution short-time-distance Fourier transform, and using the encoder as a characteristic value of audio data and a model input item; a time sequence convolutional network TCN training model is introduced, and an encoder is decoded to obtain a voice enhancement processing result; according to the method, the effect of speech enhancement is analyzed according to a previously established measurement standard, and large-scale complex audio enhancement processing is quickly and effectively performed by introducing the time sequence convolutional network. The structure, the number of layers and parameters of the time sequence convolutional network can be learned and adjusted according to actual conditions, so that the model has flexibility and adaptability, the performance and the effect of the model are further improved, and stronger support is provided for realizing large-scale multi-type speech enhancement.

The invention discloses a speech noise reduction system in a multi-noise environment, and specifically relates to the technical field of speech noise reduction, comprising a basic division module that divides speech data into a plurality of basic speech segments of the same duration; a rationality analysis module that determines whether the segmentation points of adjacent segments are reasonable, and if not, an optimization marking module that finds new segmentation points for optimization; a loop module that repeatedly verifies the optimization result until all speech segments are segmented reasonably; a noise reduction module that extracts speech and noise features, and uses a noise reduction strategy to select a model for noise reduction processing; a reorganization module that reorganizes the noise-reduced speech segments into complete speech outputs; the invention divides a speech signal into a plurality of segments, and can match the most suitable noise reduction algorithm for each segment according to its noise features, and can cope with a more complex and unstable noise environment through a dynamically adapted noise reduction strategy, and ensure that the speech quality is well improved in various scenarios.

The application discloses a natural language and meta-model driven code generation system, which comprises an AI layer containing multiple intelligent agents working cooperatively, the AI layer being configured to receive natural language instructions and perform intent recognition, entity extraction, context analysis and constraint verification through the intelligent agents to generate structured metadata; a meta-model engine configured to construct and instantiate one or more meta-model objects according to the structured metadata generated by the AI layer; and a code template engine configured to match one or more corresponding code templates from a code template library according to the content and type of the meta-model objects, and generate executable code or configuration files in one or more target languages through rendering. According to the scheme, natural language is converted into a meta-model decoupled from technology through AI, and high-quality standardized code is automatically generated through a template engine, thereby improving development efficiency and enabling developers to focus on core business innovation.

This application relates to a method, apparatus, device, medium, and product for generating visualization results of an indicator. The method includes: acquiring a data query file and a visualization conversion file; the data query file includes target library table information for each indicator among at least one indicator, the target library table information indicating a library table used to store indicator data for each indicator; the visualization conversion file includes at least one conversion file; in response to an indicator query instruction carrying an indicator to be queried, determining the target library table information of the indicator to be queried based on the data query file, and determining a target conversion file matching the indicator to be queried from the at least one conversion file; and generating a visualization result of the indicator to be queried based on the target library table information and the target conversion file. This method can improve the efficiency of generating visualization results for the indicator to be queried.

The invention discloses a hearing aid and spoken language partner training integration method and system based on hearing loss and compensation, and belongs to the crossing field of hearing aid and artificial intelligence language learning. The method is cooperatively executed by a Bluetooth headset and a mobile terminal application program, and comprises the following steps: establishing a personalized hearing loss curve of a user; the equipment has an intelligent hearing aid mode and a spoken language partner training mode. And in the hearing aid mode, real-time compensation playing is carried out on the environment sound based on the curve. In the partner training mode, three partner training processes of real-time, preset scene and word generation scene are provided, and adaptation processing is performed on the generated partner training audio based on the same hearing curve, so that key teaching phonemes are ensured to be clear and audible. The core innovation lies in that the system dynamically adjusts the compensation parameter of the hearing loss curve by analyzing the pronunciation accuracy of the user in partner training, and realizes collaborative optimization and personalized improvement of the hearing aid effect and the spoken language ability.

The invention relates to the technical field of software engineering, in particular to a code detection method and device, electronic equipment and a storage medium, and aims to improve code detection efficiency and role perceptivity. Comprising the steps of obtaining a to-be-detected code and structured data of the to-be-detected code; the structured data is used for describing the function of the to-be-detected code; based on the task splitting agent and a plurality of preset detection angles, performing task splitting on the changed sub-codes and the structured data in the to-be-detected codes to obtain a plurality of detection sub-tasks corresponding to different detection angles; for any detection angle, executing a detection sub-task corresponding to any detection angle through a detection intelligent agent corresponding to any detection angle to obtain a detection sub-result corresponding to any detection angle; and inputting the plurality of detection sub-results and the information of the target detection role into the security field large model to obtain a target detection result of the target detection role corresponding to the to-be-detected code.

The application discloses a language decoding method and device, electronic equipment and storage medium. The method comprises the following steps: obtaining an electroencephalogram decoding result of a current time step, wherein the electroencephalogram decoding result comprises electroencephalogram decoding scores corresponding to each preset syllable; expanding a historical candidate sentence of a previous time step based on candidate words associated with the preset syllable to obtain a corresponding expanded sentence set; for each expanded sentence, determining an integrated language decoding score of the expanded sentence based on a historical integrated language decoding score of the previous time step, a first language probability score of the expanded sentence and the corresponding electroencephalogram decoding score; determining a candidate sentence sequence of the current time step based on the integrated language decoding scores of the expanded sentences; performing sentence end detection based on the candidate sentence sequence of the current time step, and determining language content to be output in the current time step according to a detection result of the sentence end detection. The application improves the accuracy, intelligibility and naturalness of interaction of the output language content.

The invention provides a PDMA-based ultra-long distance speech enhancement system, which comprises a hardware acquisition end and a signal processing end, and is characterized in that the hardware acquisition end comprises a parabolic reflector and a differential microphone array, the hardware acquisition end is used for acquiring an original mixed signal, the original signal passes through the parabolic reflector to obtain a pre-enhanced signal, and the pre-enhanced signal is used for processing the pre-enhanced signal; an original signal passes through the differential microphone array to obtain a differential signal, the obtained pre-enhanced signal and the differential signal are input to the signal processing end, and the signal processing end carries out fusion processing on the received pre-enhanced signal and the differential signal to realize secondary enhancement of a target and output a target voice signal. According to the invention, the voice signal can be effectively enhanced in an ultra-long distance scene of more than 20 meters, the voice quality and intelligibility can be obviously improved, the requirements of daily communication transmission are met, the complexity of the system is reduced, the declaration cost is controlled, and the performance of the system is improved.

The present invention discloses a method, device, equipment and readable storage medium for calculating a coordinate transformation matrix, the method comprising: obtaining first coordinate data of a feature point of a target object in a reference coordinate system and second coordinate data in a coordinate system to be calibrated; establishing a coordinate transformation equation according to the first coordinate data and the second coordinate data, the coordinate transformation equation comprising a rotation matrix and a translation vector; the rotation matrix comprises a rotation angle to be optimized, and the translation vector comprises a translation parameter to be optimized; establishing an objective function according to the coordinate transformation equation; iteratively calculating the objective function according to an initial value of the rotation angle to be optimized, an initial value of the translation parameter to be optimized, and an initial value of the iteration parameter until an iteration termination condition is met, and obtaining a first updated value of the rotation angle to be optimized and a second updated value of the translation parameter to be optimized; and calculating the coordinate transformation matrix according to the first updated value of the rotation angle to be optimized and the second updated value of the translation parameter to be optimized.

The invention discloses an autism database sharing platform based on an ETL process, and belongs to the technical field of information technologies. The sharing platform comprises a data extraction module, a data conversion module, a data loading module, a metadata management module and an access control module. According to the method, efficient integration of multi-source data is achieved through the ETL process, the problems of data dispersion and format inconsistency are solved, metadata management and data standardization technologies are adopted, the understandability and utilization efficiency of the data are improved, the data security and privacy protection module is adopted to perform privacy protection on the data, and the data security and privacy protection efficiency is improved. A safe and reliable data sharing environment is established; the system can support and promote data communication and cooperation between researchers and professionals.

The application belongs to the technical field of emergency help equipment, and provides an emergency help signal panoramic projection device, method and terminal, the device can continuously detect the spatial posture and rotating motion state of itself through the environment positioning module, generate corresponding posture data and direction data, the control module can obtain the data and determine the real-time projection pointing angle of the device, and then when the device rotates, the projection control instruction corresponding to the real-time projection pointing angle is generated and sent to the image projection module, so that the image generation unit in the image projection module drives the corresponding rotating transformation of the help image, and the light source in the image projection module irradiates the help image to project the visual help signal which is right and complete. The scheme realizes omnidirectional controllable intelligent projection, can make the help signal cover all directions, at the same time ensures that the projected image can be clearly observed, is more easily identified by the rescuer, reduces the delay caused by signal blur, and improves the probability of being rescued by the help seeker.

A method for predicting pedestrian flow based on an integrated graph convolutional network with multimodal fusion, which relates to the field of intelligent traffic management and control technology. S1. Acquire and process the pedestrian flow heat map data and environmental data in hot spots, and splice the heat map data with the environmental data to achieve multimodal fusion; S2. Establish a fully connected graph network in the hot spots to obtain an adjacency matrix; S3. Design an integrated graph convolutional network based on multimodal fusion as a pedestrian flow prediction model; S4. Use the heat map features of the next moment as the label of the previous moment, train the model, and adjust the model parameters; S5. Use the trained model to predict the pedestrian flow heat map at future moments, and display the prediction results in a visual way. Using the pedestrian flow heat map data and environmental data at historical moments, predict the pedestrian flow heat map data at future moments, and display the prediction results in a visual way, the prediction results are more accurate and intuitive.

The invention discloses a method, a system, a terminal and a medium for constructing a reasoning path based on decomposition of an SQL (Structured Query Language), and the method comprises the following steps: analyzing an original SQL, and identifying detachable conditions in the original SQL to obtain a condition set, the conditions being clauses in the original SQL; deleting a plurality of conditions in the condition set on the basis of a preset deletion rule, obtaining a subSQL after deleting each condition, and obtaining a subSQL set corresponding to the original SQL; and on the basis of conditions contained in each subSQL in the subSQL set, constructing a reasoning path respectively. According to the method, the complex SQL can be analyzed into the SQL clauses and a plurality of subSQL sets containing partial conditions, and the reasoning path is constructed by analyzing the conditions contained in the subSQL sets, so that the understandability and the execution efficiency of the SQL are improved.