725 results about "Voice communication" patented technology

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 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 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, 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.

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.

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.

The embodiment of the invention provides a signal processing method and device, a model training method and device, equipment and a storage medium. The method comprises the steps of obtaining a mixed sound signal and a reference sound signal; filtering a linear echo component in the mixed sound signal based on the reference sound signal to obtain a first filtered signal; obtaining an output signal through a pre-trained nonlinear filtering model and the first filtering signal, and sending the output signal to the far-end terminal equipment; wherein the output signal filters out a nonlinear echo component of a target energy quantity level relative to the first filtering signal, the target energy quantity level is smaller than a preset energy level, and the preset energy level is determined based on the energy level of the human voice component in the first filtering signal. The nonlinear component in the sample filtering signal is filtered based on the nonlinear filtering model, and meanwhile, the filtering amount of the nonlinear component is controlled, so that excessive suppression of human voice components is avoided, and the voice communication quality in a complex environment is improved.

The present application relates to the technical field of artificial intelligence, and provides a voice communication quality evaluation method and device, a server and a storage medium. The method comprises: processing a voice segment to be evaluated of a target user to obtain a first embedded feature vector; acquiring a second embedded feature vector corresponding to a clean voice segment of the target user; concatenating the first embedded feature vector and the second embedded feature vector to obtain a third embedded feature vector; and calling a voice communication quality evaluation model to process the third embedded feature vector to obtain the MOS of the voice segment to be evaluated. According to the present application, a first embedded feature vector corresponding to a voice segment to be evaluated of a target user and a second embedded feature vector corresponding to a clean voice segment of the target user are concatenated, and then a voice communication quality evaluation model is called to process the third embedded feature vector obtained by the concatenation, thereby implementing evaluation of the voice communication quality in audio and video communication scenarios.

The invention provides a neural speech compression method and system, and the method comprises the steps: obtaining original speech data, and carrying out the encoding of the original speech data through an encoder, and obtaining a hidden vector; inputting the hidden vector into a layered residual vector quantization module, and quantizing the hidden vector into a discrete code based on a multi-scale parallel token stream; inputting the discretized code into a decoder for decoding so as to reconstruct a language signal and obtain a target language signal; wherein the encoder is configured to introduce a cross-modal distillation technology to express high-level semantics extracted from a teacher network model when the original voice data is encoded; the decoder is configured as a neural decoder that performs adversarial training enhancement with the multi-scale discriminator. According to the method, the potential structure of the signal can be learned by utilizing deep learning, breakthrough progress is achieved in a low-bit-rate scene, the modeling capability and the compression quality are improved, and the method can be widely applied to scenes such as voice communication, audio storage and edge intelligent equipment.

A signal processing method, a model training method, an apparatus, a device, and a storage medium. The signal processing method comprises: acquiring a mixed sound signal and a reference sound signal (S101); on the basis of the reference sound signal, filtering a linear echo component in the mixed sound signal to obtain a first filtered signal (S102); and obtaining an output signal by means of a pre-trained nonlinear filtering model and the first filtered signal, and sending the output signal to a remote terminal device, wherein for the output signal, a nonlinear echo component of a target energy level is filtered out relative to the first filtered signal, the target energy level is less than a preset energy level, and the preset energy level is determined on the basis of the energy level of a human voice component in the first filtered signal (S103).A nonlinear component in a sample filtered signal is filtered out on the basis of a nonlinear filtering model, and the amount of the nonlinear component filtered out is controlled, avoiding over-suppression of the human voice component, and improving the voice communication quality in complex environments.

An apparatus for audio communication includes a memory configured to receive audio data from a user of a voice communication, and one or more processors in communication with the memory. The one or more processors are configured to receive the audio data for the voice communication, which includes voice data of the user and background audio data. The processors are further configured to classify the background audio data into a plurality of types of background audio data, determine to not suppress a subset of the plurality of types of background audio data, process the audio data to not suppress the subset of the plurality of types of background audio data to generate output audio data, and transmit the output audio data.

The invention discloses a psychological counseling system and method based on an AI model and voice communication. The psychological counseling system comprises an interactive voice response module; a high risk monitoring module; a psychological counseling AI component; a manual consultation module; an information acquisition module; and a historical dialogue data acquisition module. The psychological counseling method and the psychological counseling device solve the problems that in an existing psychological counseling process, a user cannot conveniently use voice and an AI model to carry out dialogue, a channel for people to obtain psychological health support is limited, and the authenticity and instantaneity of interaction are reduced.

The invention relates to the technical field of voice communication, and discloses a frequency domain noise reduction and reduction method and system for double-group human voice synchronous acquisition, and the method comprises the steps: obtaining a synchronous signal through double groups of acquisition points, constructing a double-source frequency domain coordinate system, and mapping the coordinate system into an energy distribution point; calculating a spatial relationship to identify an emotional distortion state, and dividing four quadrants to generate a distortion type mark; dividing a frequency domain signal and executing targeted reconstruction and correction, and finally dividing a frequency band enhanced energy remarkably-reduced part; according to the method, the voice composite distortion under the extreme emotion can be accurately processed, and reliable voice quality guarantee is provided for accurate transmission of key information in an emergency communication scene.

The invention provides an extremely low rate voice communication method and related equipment. The method comprises the following steps: extracting a first feature and a second feature from acquired voice information; the first feature reflects semantic information of the voice information; the second feature reflects the acoustic feature of the voice information; calculating an entropy value of variational posteriori distribution of the first feature, and adjusting the first feature according to the entropy value to obtain a third feature; performing semantic quantization on the third feature to obtain a first index; performing acoustic quantization on the second feature to obtain a second index; and gradually restoring details of the voice information based on the first index and the second index so as to complete transmission of the voice information. According to the embodiment of the invention, through semantic and acoustic feature fusion and entropy modulation and quantization, the compression efficiency is improved, the bit demand is reduced, and high-quality voice transmission and restoration are realized.

The invention provides an interactive voice response processing method and device, equipment and a storage medium, and relates to the technical field of voice communication, and the method comprises the steps: obtaining IVR process configuration information related to an incoming call request under the condition that the incoming call request of a client is received through a telephone soft switch platform, iVR branch process configuration information in the IVR process configuration information comprises information of a plurality of nodes and execution conditions among the nodes; executing a starting node in the IVR process configuration information, and searching a to-be-executed node according to an execution condition between nodes in the IVR branch process configuration information related to the starting node; the to-be-executed node is executed, the next execution node of the to-be-executed node is searched according to the execution condition between the nodes in the IVR branch process configuration information related to the to-be-executed node and / or the key information of the client responding to the to-be-executed node, and the scheme of the invention can improve the IVR processing efficiency and improve the user experience.

Aspects of the subject disclosure may include, for example, analyzing, by a processing system including a processor, at least first data from a voice communication session, generating, by the processing system and based on the analyzing, a prediction that second data associated with the voice communication session is likely to experience degradation in an amount greater than a threshold, identifying, by the processing system and based on the generating, a source of the degradation, and implementing, by the processing system and based on the identifying, a modification to at least one of a user equipment or a network resource. Other embodiments are disclosed.

The invention relates to the technical field of voice communication, and discloses a voice communication method based on an LDSW voice compression technology and a smart phone, and the method comprises the steps: enabling an LDSW smart phone to achieve the low-power-consumption digital communication with other LDSW smart phones through an LDSW smart label by adopting an LDSW communication technology standard, and enabling the LDSW smart phone to achieve the low-power-consumption digital communication without relying on a mobile network; when voice communication is carried out between the LDSW smart phones, firstly, input voice is subjected to structural compression at a transmitting end, and the voice is converted into voice text information and then is transmitted to a receiving party through the LDSW smart tag; and after receiving the voice text information, the receiver performs voice synthesis on the voice text information and other information required for voice restoration, and then restores and plays the voice input by the transmitting end. By integrating a conventional smart phone and an LDSW smart power bank and combining a structured voice compression and peacetime and urgency mode switching method, daily functions of equipment are reserved, dependence on a public network is not needed in emergency, confidential voice communication can be realized with low power consumption, and personalized voice can be restored.

A voice communications system comprises a user interface with a push-to-talk (PTT) button, a communication module configured to transmit a signal from a first user to a second user upon activation of the PTT button without establishing a communication channel, and a receiver module that allows the second user to either accept the signal, thereby initiating a live audio stream, or ignore it. The system includes a voice streaming module for transmitting audio data while the PTT button is engaged and an artificial intelligence filtering module for real-time classification and moderation of the audio data, ensuring that the audio data is sent to the recipient only after approval by the filtering module. Additionally, the system can moderate pre-recorded digital messages prior to their transmission to a recipient.

A user interface system enables communication through inaudible speech by detecting physiological signals associated with inner voice communication. The system includes a wearable device positioned in, on, or around a user's ear, housing one or more sensor modalities such as electromyography (EMG) sensors for detecting muscle movements, functional near-infrared spectroscopy (fNIR) sensors for brain activity, and / or detection-and-ranging systems using sonar or radar for micro-deformations. Pre-processing modules condition the sensor data, which is then analyzed by one or more machine learning models to reconstruct the content of the user's inaudible communication and produce output representations. The system enables high-bandwidth natural language interaction without audible vocalization, maintaining privacy while supporting applications including AI assistant interaction, real-time translation, secure authentication, and inter-party communication.

The invention discloses a voice interaction method and device and electronic equipment. The method comprises the following steps: identifying service data sent by a terminal to obtain an identification result; under the condition that the identification result indicates that the service data is the voice service, semi-persistent scheduling (SPS) configuration information is sent to the terminal through radio resource control (RRC) signaling, and SPS configuration corresponding to the SPS configuration information is activated through control signaling; and after the SPS configuration is successfully activated, voice data sent by the terminal according to the SPS configuration information is received through the satellite, and data transmission optimization is carried out according to the propagation characteristics of the satellite in the voice data receiving process. According to the method and the device, the technical problems of low voice communication quality and stability caused by the fact that the standard native voice service in the IoT NTN is not supported and complex propagation characteristics in a high-orbit satellite communication environment are solved.