896 results about "Voice communication" patented technology

The invention discloses a commerce and trade circulation supply chain optimization method based on multi-modal data fusion, and the method comprises the following steps: 1, multi-modal data collection: collecting the multi-modal data of the whole supply chain in real time through an Internet of Things device, an enterprise ERP system, a logistics tracking system and a third-party platform, the multi-modal data comprises text data, image data, voice data, time sequence data and spatial data. When the scheme is implemented, multi-source heterogeneous data of the Internet of Things, ERP and the like are integrated, so that the problem of an information island in a traditional supply chain is effectively solved; position coding and a cross-modal attention mechanism are introduced into the multi-modal fusion model, so that the data alignment precision is remarkably improved, for example, when'emergency order adding 'in a voice communication record is associated with a corresponding order image, the weight distribution accuracy is improved; the application of the block chain technology ensures the transparency and traceability of the supply chain decision.

The disclosed invention provides a system and method for processing air traffic control (ATC) communications to ensure real-time validation of aircraft operations and improve pilot logbook accuracy. The system receives ATC voice communications through a digital radio system, converts them to text using a natural language processing model trained on aviation terminology, and extracts key operational data, including aircraft identifiers, instructions, and corrections. This data is correlated with Automatic Dependent Surveillance-Broadcast (ADS-B) and GPS data to verify compliance with ATC instructions. The system creates a continuous transcript of ATC-pilot communications, stitching together conversations across multiple frequencies. Deviations from ATC instructions are detected, generating alerts and logging the events. The method further provides automated reporting, archiving of communications, and enhanced safety metrics in immutable pilot logbooks, using blockchain technology for secure storage. This invention improves operational safety, transparency, and flight data integrity.

The invention relates to the field of intelligent agriculture and intelligent construction, and discloses an agricultural monitoring system and method based on an intelligent helmet, and the system comprises a wearable terminal, an active sampling control module, an image recognition processing module, a multi-modal fusion module, a recognition reasoning module, and a result output module. The method comprises the steps of collecting a field crop image and environment data; calculating an uncertainty index of an identification result according to the current image; if the uncertainty is high, adjusting the acquisition angle of the camera and re-sampling; constructing the image and the agricultural environment data into a multi-modal graph structure and carrying out graph neural network processing; performing dynamic trend identification on a fusion result, and outputting a crop state; and outputting an identification result through a helmet display or linking an agricultural control system to execute related operations. According to the invention, the capabilities of identification display, voice communication, operation guidance and system integration in a complex environment are improved, and the problems of poor real-time performance and weak interaction of an existing scheme are solved.

The invention discloses a talkback method and system based on label door lock cooperation, and the method comprises the steps: broadcasting door lock binding information: obtaining a broadcasting result of the door lock binding information according to the fact that a door lock A detects that a Bluetooth label Tag-DID carried by an owner a leaves a communication range; neighbor door lock state detection: sending an empty query request according to a call button of a Bluetooth tag pressed by an owner a, and obtaining a response result of an idle door lock; door lock selection and call establishment: selecting a door lock ID displayed on a display screen of a door lock Z according to the owner a, extracting the cached IP by the door lock Z and initiating a call request to the door lock A by the door lock Z to obtain an establishment result of call connection between the door locks; and voice communication and switching management: realizing voice communication between the door lock Z and the door lock A according to the established call connection, and obtaining a cross-door-lock voice communication switching result. By utilizing the embodiment of the invention, the cooperative work capability between the door locks can be improved, and the communication experience of a user in an intelligent home environment is optimized.

The invention discloses an AI-based illumination control system self-adaptive dimming method, which comprises the following steps of: acquiring a two-dimensional coordinate, a motion track curve and voice communication data of a personnel position, and identifying social interaction scene categories of different time slices; determining a social interaction strength score according to the voice communication data, and predicting the most adaptive spectrum configuration data in the current social interaction scene in combination with the social influence factor matrix and the social interaction scene type; according to a historical dimming log and an energy consumption metering curve, identifying a short-term high-frequency dimming segment, and adjusting a real-time response strategy and a disturbance classification processing mechanism of the spectrum configuration data; and adjusting the illumination power distribution values of the high-frequency use area and the low-frequency use area according to the occupation raster data generated by the millimeter-wave radar and the infrared array in combination with the lamp area mapping relation and the power budget constraint. Through multi-source data fusion and a self-adaptive regulation and control mechanism, unification of real-time performance, pertinence and energy-saving performance of spectrum regulation is realized, and the dynamic response capability and the user perception experience of the intelligent illumination system are improved.

The invention provides an emotion-synchronized 2D digital human model training method and device, and relates to the technical field of 2D digital humans. According to the method, an audio and video data set containing various expressions and emotions is generated, a LivePortrait model, a sound cloning technology and a hallo2 model are utilized, accurate alignment of a digital population type and an input audio is achieved, and the emotion-synchronized 2D digital human model is obtained. And meanwhile, the facial expression consistent with the audio emotion is kept. Meanwhile, an emotion classifier is used in a model training stage, so that the reasoning speed is increased; an expression loss function and a mouth loss function are designed, the model training process is optimized, and it is ensured that the expression and mouth of the digital human can accurately reflect the emotion of the input audio. The method aims at improving the expression naturalness and reality sense of a 2D digital person in voice communication and ensuring that the expression of the digital person is synchronous with the emotion in audio.

The invention discloses a cross-language voice communication system fusing AI voice cloning and real-time translation, and relates to the technical field of voice processing, and the system comprises the steps: obtaining a first input voice of a first client, and carrying out the language type recognition to obtain a first input voice language type; when the language type of the first input voice is different from the language type identifier of the second client, performing semantic recognition to obtain a character recognition result of the first input voice; translating the first input voice character recognition result according to the second client language type identifier to obtain a corresponding voice character recognition result; obtaining physical feature information of the first input voice; and performing voice cloning according to the physical feature information and the character recognition result of the second input voice to obtain the second input voice, and sharing the second input voice to the second client. According to the invention, the technical problems of lack of personalized voice feature reservation and insufficient real-time performance in cross-language communication in the prior art are solved, and the technical effect of cross-language and personalized real-time voice communication is realized.

The invention provides a game resource optimization method and device based on a predictive algorithm and a storage medium. The method comprises the following steps: compiling a source code of a game voice tool into a bytecode file and generating an executable game voice tool main body; dividing the loading process of the non-core resources into a plurality of stages, and pre-loading the non-core resources through at least one background thread; generating a target resource list for a subsequent game scene by using a predictive algorithm; executing a loading operation on the target resource which is not loaded by utilizing a background thread, and executing a releasing operation on the loaded resource which is judged to be lower than a preset threshold value in the subsequent game scene by the predictive algorithm, wherein the use probability of the loaded resource is lower than the preset threshold value; a high execution priority is set for a voice communication module, and the loading priority of non-core resources and the task allocation order of background threads are dynamically adjusted according to the real-time condition of voice communication. According to the method and the device, memory resource utilization can be improved, CPU (Central Processing Unit) occupation is reduced, and game jamming caused by resource competition is reduced.

The invention discloses a mobile intelligent safety supervision device based on multi-mode perception and edge calculation, and the device comprises a camera which is a high-performance dual-camera with adjustable height and visual angle, is fixed on a telescopic aluminum alloy support, integrates a loudspeaker and a voice communication function, and supports the autonomous network and 4G bidirectional communication; the edge intelligent gateway is used as a core computing unit, supports a neural network algorithm to analyze audio and video data, and realizes real-time identification of security behaviors; the Beidou positioning module is integrated on the edge intelligent gateway and provides centimeter-level high-precision positioning; the data storage module is used for large-capacity data storage and supporting storage and retrieval of audio and video streams and analysis data; the network transmission module is a multi-protocol network transmission module and supports WiFi, 4G and wired network transmission; and the power supply module is high in capacity and adopts a pluggable battery design. Omnibearing high-definition video monitoring can be realized, and the requirements of real-time communication and real-time early warning between project responsible personnel and on-site construction personnel are met.

The invention discloses a voice optimization method and device for internet protocol voice communication, equipment, a storage medium and a program product, and relates to the technical field of data processing, and the voice optimization method for internet protocol voice communication comprises the steps: obtaining network features and voice data of a voice communication network; performing network jitter prediction based on network features, and dynamically adjusting an adaptive jitter buffer based on a prediction result; performing data enhancement processing based on the voice data to obtain enhanced voice data; caching enhanced voice data based on the dynamically adjusted adaptive jitter buffer area; and encoding and decoding the enhanced voice data in the adaptive jitter buffer area to obtain optimized voice data. Through processing of network characteristics and voice data, voice optimization of Internet protocol voice communication can be carried out more comprehensively. By adjusting the adaptive jitter buffer area, high adaptation to a complex network environment is realized, and the quality of voice communication is effectively guaranteed.

The invention provides a position state interaction method and system between interphones and related equipment, and the method comprises the steps: obtaining the first position state information of a local interphone in response to a position information updating request of networking equipment; modulating the first position state information, and transmitting the modulated first position state information to a surrounding networking interphone through radio frequency; receiving a radio frequency signal containing second position state information fed back by the surrounding networking interphone based on the automatic position return system; and generating an equipment list and a relative position map of the local interphone and the surrounding interphones in a first interface of the local interphone based on the first position state information and the second position state information. According to the technical scheme, the positions of the interphones can be automatically exchanged through the automatic position return system, the equipment and the positions of the interphones around can be displayed on the first interface of the interphone, and the technical problem that position interaction between the interphones can only be performed through voice communication and is very inconvenient is solved.

Systems and methods are provided for performing voice communication operations. The system establishes, by a first augmented reality (AR) device, a voice communication session between a plurality of users. The system displays, by the first AR device of a first user of the plurality of users, an avatar representing a second user of the plurality of users. The system receives, by the first AR device of a first user of the plurality of users, input from the first user that selects a display position for the avatar representing the second user within a real-world environment of the first user. The system animates the avatar representing the second user based on movement information received from a second AR device of the second user.

The invention discloses a voice packet loss processing method and device based on deep learning and variational mode decomposition, equipment and a storage medium, and relates to the technical field of communication, and the method comprises the steps: extracting the multi-dimensional features of a received voice signal; performing packet loss detection on the multi-dimensional features by using a space-time convolution automatic encoder network to generate estimation features of lost frames, and reconstructing the voice signal based on the estimation features to obtain the voice signal after interpolation reconstruction; performing variational mode decomposition on the speech signal after interpolation reconstruction to obtain an effective component; according to the method, the effective component is reconstructed to obtain a filtered voice signal, post-processing optimization is carried out on the filtered voice signal to obtain an optimized voice signal, and by fusing the feature learning capability of deep learning and the adaptive signal processing advantage of variational mode decomposition, bandwidth consumption is reduced, voice communication quality in a packet loss scene is improved, and voice communication quality is improved. And the requirements of real-time, high-efficiency and high-quality VoIP communication are met.

The invention relates to the technical field of voice communication, and provides an online conference transcription method capable of realizing intelligent conference summary, which comprises the following steps: collecting audio data of each speaker in a conference process in real time through a microphone array, and carrying out primary processing on the collected audio signals; performing sound source positioning on the audio signal and determining a source direction of the audio signal according to a sound source positioning result; performing voiceprint recognition on the audio signal to extract voice spectrum features of the speaker, and marking the voiceprint of the speaker according to a voiceprint recognition result; and the weighted audio signal is used for character transcription, and the speaker is positioned to generate a complete conference summary. The problems that in a traditional audio transfer system, multiple speakers cannot be accurately recognized, speech content cannot be distinguished, and manual proofreading work cannot be reduced are effectively solved, and the accuracy and efficiency of conference recording are improved.

The invention relates to the technical field of voice communication, in particular to a voice signal processing method and device, equipment, a storage medium and a computer program product. The method comprises the following steps: performing real-time noise classification on a received original voice signal based on a preset deep neural network noise recognition model; according to the noise classification result, selecting a corresponding combined noise reduction model to perform multi-stage noise reduction processing on the original voice signal to obtain a noise-reduced voice signal, the combined noise reduction model comprising one or more of a deep neural network-based noise reduction module, a Wiener filtering module and a spectral subtraction module; and on the basis of time domain feature extraction and frequency domain feature extraction, voice feature enhancement processing is performed on the noise-reduced voice signal to obtain the target voice signal, so that the transmission quality of the voice signal is improved.

A system is presented that includes secure push-to-talk voice functionality. Using encryption, authentication, user filtering, and integration with new and existing LMR systems, a secure voice platform ensures malicious software, unauthorized access and brute force security attacks will not compromise the voice communications of the system. The VoIP system is engineered to ensure graceful system degradation in the event of maintenance activities, natural disasters and failure modes. The hardened VoIP system offers the functions a LMR trunking system while utilizing broadband connections. Private calls, group calls, Emergency Alarms with covert monitoring capability, scanning and priority scanning may be incorporated into the system. The system includes a VoIP-controller that serves as a trunking controller, manages available VoIP-based conference bridges, and assigns them as needed to the parties involved in each voice call. The system includes multiple fallback methods that may be prioritized based on pre-failure analytics.

The invention relates to the technical field of medical software, in particular to a system and method capable of recognizing early Alzheimer's disease through mobile phone voice, the system capable of recognizing early Alzheimer's disease through mobile phone voice comprises intelligent mobile phone software and a hospital background, and the mobile phone software builds an interactive platform based on the hospital background. Through intervention monitoring of mobile phone software, when voice communication, chatting and instruction input are carried out in daily life, monitoring can be carried out through system monitoring of software and characteristics such as smoothness and continuity of voice, so that analysis and diagnosis of the Alzheimer's disease can be carried out according to comparison with a normal state, and moreover, in the monitoring process, the monitoring efficiency is improved. Diversified collection is carried out in different time periods and different statement numbers, language recognition and dialect recognition are added when a user inputs voice, and keyword recognition for determining the Alzheimer's disease condition can be carried out through approximate comparison with mandarin after the user speaks out statements.

The invention discloses a voice QoS optimizing and playing method, device and equipment for end-to-end information source encryption and a medium, and relates to the field of instant messaging, and the method comprises the steps: storing a received encrypted voice data packet into a jitter buffer area in a ciphertext form; determining a current target buffer water level of the jitter buffer based on the current network jitter degree and a delay factor introduced by the current decryption processing; generating a play control strategy and a decryption scheduling strategy according to the current actual buffer water level and the current target buffer water level of the jitter buffer; the decryption scheduling strategy comprises a decryption operation frequency, a triggering opportunity and an execution priority, and the decryption scheduling strategy is adjusted based on a real-time playing control strategy and a jitter buffer area state; and scheduling the corresponding encrypted voice data packet from the jitter buffer area by using a decryption scheduling strategy for decryption, and decoding and playing the decrypted voice data packet according to a playing control strategy. According to the invention, the overall fluency of encrypted real-time voice communication is improved.

The invention relates to the technical field of interphone communication, and discloses an interphone communication method based on a base station type relay, a base station, an interphone and a medium, and the method comprises the steps: transmitting a time frame to a network node in a first communication range based on a first voice communication channel, network nodes corresponding to all voice time slots in a time frame are broadcasted through a synchronous time slot of the time frame, the time frame comprises the synchronous time slot, a competition time slot and at least two voice time slots, and each interphone serves as one network node; acquiring voice information transmitted by the network node and other base stations through the voice time slot of the received time frame; broadcasting the received voice information to network nodes in a first communication range based on a first voice communication channel, and broadcasting the received voice information to other base stations in a second communication range based on a second voice communication channel, according to the invention, the mutual full-duplex talkback communication between the base station and the interphone in the whole large network can be realized, the communication time delay is short, the movable networking can be realized, and the networking is flexible.

The invention discloses a communication method and device, electronic equipment and a readable storage medium, and belongs to the technical field of communication. The method comprises the following steps: under the condition that a voice call is carried out between first electronic equipment and second electronic equipment through a first communication link, the first electronic equipment acquires first information, the first information comprises at least one of the following items: network state information of the first electronic equipment, environment information of an environment where the first electronic equipment is located, and a motion state of the first electronic equipment; the first electronic equipment establishes a second communication link with the second electronic equipment based on the first information; and the first electronic equipment performs voice communication with the second electronic equipment through the second communication link or the first communication link and the second communication link.

The invention discloses a fire emergency response capability assessment method based on artificial intelligence, and the method employs an AI algorithm to automatically analyze and calculate assessment data, eliminates the subjectivity of manual assessment, and guarantees the objectivity and consistency of an assessment result. The performance of the trainee is monitored and evaluated in real time, so that feedback can be timely obtained and adjustment can be carried out in drilling and actual combat, the emergency response effect is improved, and the pertinence of training and drilling of the trainee in the later period is enhanced; according to the scheme, multiple factors such as physiological performance, action performance, voice communication and team cooperation are comprehensively considered, and the emergency response ability of individuals and teams is quantitatively evaluated from multiple dimensions; the comprehensive evaluation can reflect the real ability of the trainee more comprehensively, and provides a basis for personalized training; by introducing the concept of task matching degree, the evaluation result not only can reflect the individual ability, but also can perform adaptive adjustment for specific tasks and roles, thereby being beneficial to optimizing role allocation and improving the overall efficiency of a team.

This invention provides a new and improved voice communication system with high quality noise cancellation method and devices to overcome the limitations and difficulties encountered in conventional technologies. This invention discloses a noise cancellation apparatus that includes a vibration sensor and a microphone for receiving and transmitting voice signals as incoming speeches. The vibration sensor is applied to receive vibration signals corresponding to the voice signals for applying the vibration signals as reference signals for removing noise signals generated from environmental noises by converting vibration signals to intermediate PDL representation together with the speaker characteristics, mapping them into full band high quality clean acoustic representation, and synthesizing clear personal speech with characteristics identical to the original microphone speech without noises.

A voice masking method, device and system are suitable for the field of intelligent vehicles, and the method comprises the following steps: determining a voice source position and a target masking position; receiving a sound signal from the voice source position, and detecting whether a voice signal exists in the sound signal so as to generate a detection result; if the detection result is that a voice signal exists in the sound signal, generating masking sound for the voice signal, and outputting the masking sound to a first loudspeaker at the voice source position and a second loudspeaker at the target masking position; and if the detection result is that the voice signal does not exist in the sound signal, no masking sound is generated. According to the invention, the requirement of passengers for private voice communication in the vehicle cabin space can be met, and the information safety is improved.

The invention provides an unmanned aerial vehicle-based traffic accident auxiliary processing method and system, and the method comprises the steps: determining a first coordinate of an accident site on an electronic map according to alarm information, and obtaining second coordinates and electric quantity of a plurality of unmanned aerial vehicles in a standby state at the periphery; respectively calculating the distance between the unmanned aerial vehicle and the accident site, and calculating the distance score and the electric quantity score of each unmanned aerial vehicle; performing comprehensive scoring according to a preset distance weight and an electric quantity weight, and selecting the unmanned aerial vehicle with the highest comprehensive score to execute a task; the node with the minimum f (n) value is selected as the current node, and circulation is carried out in sequence until the first coordinate is reached, and a flight path is obtained; and the unmanned aerial vehicle goes to the accident site according to the flight path, shoots a field picture according to the control instruction, performs voice communication with the field, obtains picture data and voice data, and returns the picture data and the voice data to the command center. According to the invention, the most suitable unmanned aerial vehicle can be quickly and accurately selected to go to the accident scene, and the path and unmanned aerial vehicle unified scheduling are planned.

A wireless patient communication device, disclosed for hospital emergency departments, specifically for patients in hallway beds. The device enables direct communication between patients and hospital staff through a built-in speaker and microphone. When patients press the call button, the device initiates a call to a pre-programmed recipient, allowing two-way communication. An indicator light illuminates when calls are placed, remaining on until manually turned off, ensuring staff awareness. The device includes emergency buttons: A Code Blue button, which activates a flashing blue light and an emergency alert, and a Rapid Response button, which triggers a flashing red light. A swappable battery ensures uninterrupted operation, while a metal clamp allows attachment to hospital beds or IV poles. The system integrates with a central database for monitoring and logging alerts, improving patient safety and hospital efficiency.

The invention relates to the technical field of wireless communication, in particular to a public network interphone with a quantum chip. According to the interphone, a master control module and a quantum security chip work cooperatively, a quantum random number generator is used for dynamically deriving a session key, and offline security distribution of a group master key is realized by encrypting a two-dimensional code; in a communication process, a system adaptively switches a voice coding mode according to a network condition, end-to-end authentication encryption is carried out on voice data by using a hardware encryption engine, and low-delay transmission is guaranteed through a network priority mark; the device establishes a complete key life cycle management mechanism, supports regular update and forward security of session keys and instant update and backward security of a group master key when members change, and ensures that all key operations are completed in a quantum chip. According to the invention, various eavesdropping and tampering attacks are effectively resisted, and safe, real-time and clear high-confidentiality voice communication in a public network environment is realized.

The invention discloses a real-time voice communication system and method based on a movement prediction and standby path mechanism, and aims to solve the problems of voice packet loss, delay and call interruption caused by frequent link interruption, large route reconstruction delay and high control overhead when an existing outdoor dynamic small-scale team performs real-time voice communication in a high-speed movement scene. The system comprises a voice sending end, a voice receiving end, a path management module and a mobile prediction module. The path management module constructs and maintains a main path tree for data transmission and a pre-generated standby path tree; the movement prediction module indirectly predicts the movement trend of the node through the change trend of the link quality, and constructs a movement prediction quality parameter through the change of the link packet loss rate, the delay and the jitter so as to dynamically optimize the structure of the path tree; when the main path is subjected to link breakage or the communication quality, such as packet loss rate, delay and jitter deterioration, is below threshold values, the system can be seamlessly switched to the standby path tree, so that the continuity and real-time performance of voice communication are guaranteed; according to the invention, the lightweight design is adopted, the control packet overhead is reduced, the method is suitable for an outdoor real-time voice communication scene with a small number of nodes, high moving speed and limited bandwidth and power consumption, and the communication robustness and the user experience are remarkably improved.

Systems and methods detect and prevent vishing attacks through an integrated framework combining SIP header customization, STIR / SHAKEN frameworks, AI / ML analysis, and real-time speech analysis using the Viterbi algorithm. The system begins with call initiation, embedding authentication information in the SIP header. The SIP data is transmitted and verified using STIR / SHAKEN frameworks, ensuring the authenticity of the caller's identity. Verified data is cross-referenced with third-party databases and analyzed by an AI / ML engine to detect anomalies. If potential fraud is detected, the call is blocked, and the customer is notified. Calls that pass initial checks are further analyzed using the Viterbi algorithm, which converts speech to text and identifies suspicious patterns. An anomaly pattern detector processes the converted text to detect vishing indicators, terminating the call if a match is found. This multi-layered approach ensures robust protection against vishing, enhancing the security and reliability of voice communications while safeguarding users from fraud.

The invention discloses a voice processing method and device and a storage medium, and the method comprises the steps: an intelligent helmet employs a voice communication protocol of a dual-mode Bluetooth module to build a first voice channel between the intelligent helmet and a target vehicle, and transmits a captured voice signal to the target vehicle through the first voice channel; the target vehicle carries out instruction type analysis on the voice signal to obtain an instruction type result, and if the instruction type result represents that an operation corresponding to the voice signal is an operation of controlling the mobile terminal to execute a telephone service, channel switching indication information is sent to the intelligent helmet; the intelligent helmet disconnects the first voice channel according to the channel switching indication information and establishes a second voice channel with the mobile terminal; according to the invention, the voice transmission among the target vehicle, the intelligent helmet and the mobile terminal is realized, the voice interaction convenience is effectively improved, and the voice interaction method and the intelligent helmet have the advantages that the voice transmission among the target vehicle, the intelligent helmet and the mobile terminal is realized.

System, process, and device configurations are provided for nonplayer character (NPC) personalization and electronic game control. Methods can include receiving user input to personalize at least one nonplayer character (NPC) of an electronic game, generating at least one NPC seed element for the at least one NPC character, and controlling output of the electronic game with at least one NPC character presented including an NPC seed element. NPC seed elements may be determined from one or more of communication steams including messaging data, game chat and voice communications. In addition, systems and methods include training a machine learning model for generatively producing NPC communications, NPC interactions, and NPC appearance. Operations also allow for detecting electronic game data to identify NPC messaging for a game including modification of a communication style NPC character while maintain NPC game messaging.