150 results about "Voice functions" patented technology

An apparatus for virtual avatar generation is provided, including a face encoder, a face fusion engine, a voice encoder, and a voice fusion engine. The face encoder extracts first facial features from the first facial image and then encodes they into M first facial feature vectors and extracts second facial features from the second facial image and then encodes they into M second facial feature vectors. The face fusion engine generates a facial fused feature vector from the facial feature vectors, thereby generating a fused image. The voice encoder extracts first voice features from the first audio clip and then encodes they into N first voice feature vectors and extracts second voice features from the second audio clip and then encodes they into N second voice feature vectors. The voice fusion engine synthesizes a voice fused feature vector from the voice feature vectors, thereby generating a fused voice.

Systems and methods are presented herein for increasing user engagement with an interface by suggesting commands or queries for the user. A plurality of content items available for consumption are identified and metadata for each of the plurality of content items is retrieved. One or more candidate voice commands are generated based on a plurality of voice command templates based on a target verb and a subset of the metadata corresponding to the plurality of the content items available for consumption. A recall score is generated for each candidate voice command based at least in part on a detection of phonetic features that match between clauses of each candidate voice command. At least the candidate voice command with the highest recall score is selected and output using a suggestion system.

The invention provides a multi-player voice game performance optimization method and device based on dynamic resource management. The method comprises the following steps: acquiring a core source code of a game voice tool and pre-compiling the core source code to generate a byte code file; analyzing a target scene demand and pre-loading a related resource file; performing real-time acquisition and analysis on memory occupation and CPU load to obtain performance state data of the current system; based on the performance state data, when it is detected that memory occupation and / or CPU load exceed a preset threshold value, a pre-compiling mode and a resource loading strategy are automatically adjusted; when it is detected that the multi-person voice function is started, resources and byte codes needed by voice processing are placed in a high-priority queue, and cooperative scheduling is carried out on memory occupation and / or CPU loads. According to the method, the loading delay can be reduced, the frame rate can be improved, the communication delay can be reduced, and real-time voice processing and high-load requirements of games can be dynamically considered.

The utility model relates to a 5G optical fiber remote system supporting a voice function. Comprising a duplexer 1, a low noise amplifier 1, a down converter 1, an AD analog-to-digital converter 1, a power amplifier 1, an up converter 1, a DA digital-to-analog converter 1, a single chip microcomputer MCU1, a voice processing module, a duplexer 2, a low noise amplifier 2, a down converter 2, an AD analog-to-digital converter 2, a power amplifier 2, an up converter 2, a DA digital-to-analog converter 2, a digital signal processor FPGA1 and an optical module 1. The far-end unit comprises a duplexer 3, a low-noise amplifier 3, a down-converter 3, an AD analog-to-digital converter 3, a power amplifier 3, an up-converter 3, a DA digital-to-analog converter 3, a single-chip microcomputer MCU2, a duplexer 4, a low-noise amplifier 4, a down-converter 4, an AD analog-to-digital converter 4, a power amplifier 4, an up-converter 4, a DA digital-to-analog converter 4, a digital signal processor FPGA2 and an optical module 2. According to the utility model, main parameters can be modified, set and output according to voice.

A web page embeds an AI-driven widget that converts static content into an interactive experience. Executable code builds a page interaction index from the page's DOM, including text, selectors, and positional metrics for DOM nodes. In response to natural-language input, pipelines perform summarization, stepwise explanations, voice-guided form completion with rule-based validation, and on-page product scanning to create a dynamic, user-tunable comparison table. Results are rendered as in-place overlays with interactive back-references that highlight source DOM nodes in the viewport. Speech recognition and text-to-speech enable multimodal interaction. Optional privacy gating redacts sensitive data or routes processing to local models. The system improves webpage usability by binding AI outputs to precise DOM regions and providing unified, context-aware assistance within the page.

This application discloses a speech enhancement method (100, 200) and device. The speech enhancement method (100, 200) includes: receiving a current audio input signal having a speech portion and a non-speech portion (201); determining a voice feature of the speech portion in the current audio input signal (202); determining a speech quality of the current audio input signal (203); evaluating whether the speech quality meets a predetermined speech quality requirement (204); and creating or updating, in response to the speech quality meeting the predetermined speech quality requirement, a reference speech feature by using the voice feature, where the reference speech feature is used for enhancing the speech portion in an audio input signal (205).

A speech recognition method and a method for training a deep learning model are provided. The speech recognition method includes: obtaining a first speech feature of a speech to-be-recognized, which includes a plurality of speech segment features corresponding to a plurality of speech segments; decoding the first speech feature using a first decoder to obtain a plurality of first decoding results corresponding to a plurality of the words, indicating a first recognition result of words; extracting a second speech feature from the first speech feature based on first a priori information, which includes the plurality of first decoding results, and the second speech feature includes first word-level audio features corresponding to the plurality of words; and decoding the second speech feature using a second decoder to obtain a plurality of second decoding results corresponding to the plurality of words, indicating a second recognition result of the word.

The invention belongs to the technical field of intelligent household appliances, and particularly relates to an offline voice broadcast method, device and equipment and a computer storage medium. After a natural language instruction sent by a user is obtained, the natural language instruction is immediately sent to a WiFi module of the intelligent household electrical appliance, then the WiFi module of the intelligent household electrical appliance generates request information corresponding to the natural language instruction, and a feedback result corresponding to the request information is obtained through a cloud server of the intelligent household electrical appliance. And the voice module of the intelligent household electrical appliance is controlled to broadcast the broadcast corpus in the feedback result. According to the method, when the intelligent household electrical appliance with the voice technology is in an offline environment, the intelligent household electrical appliance can still use the voice function application depending on the online service, so that the intelligent household electrical appliance becomes more convenient and faster in the daily life of the user, and the use experience of the intelligent household electrical appliance is improved.

Disclosed are a multi-speaker overlapping voice detection method and a system. The method includes: obtaining a voice to be detected, and removing silence from the voice to be detected is removed; extracting a feature of the voice to be detected after silence removal to obtain a voice feature of the voice to be detected; and inputting the voice feature into an overlapping voice detection model to obtain an overlapping speaker number corresponding to the voice to be detected output by the overlapping voice detection model. The overlapping voice detection model is obtained by supervised training based on a voice feature of a sample voice and a corresponding label of the overlapping speaker number, extracts an embedding of the voice feature, and classifies the overlapping speaker number to obtain the overlapping speaker number of the voice to be detected based on the extracted speaker embedding.

The invention discloses an intelligent voice rehabilitation training management system. The system comprises an intra-hospital local area server, a preoperative risk assessment module, an intra-operative neural function assessment module, an acoustic assessment module and an individualized rehabilitation training scheme making module. The system is in butt joint with intraoperative nerve monitoring equipment to collect electromyographic signals in real time, and multi-modal fusion analysis is carried out in combination with preoperative / postoperative acoustic parameters and spectrograms. The system utilizes a deep learning model to predict the noise disorder risk and the recovery trajectory at different time points after the operation; three-dimensional training navigation is constructed based on postoperative time, neurological function parameter changes and vocational voice requirements, a rehabilitation path is dynamically optimized by adopting near-end strategy optimization reinforcement learning, and training intensity and content are adjusted in a closed-loop mode through auditory feedback; closed-loop management of preoperative evaluation, intraoperative monitoring, postoperative evaluation, individualized rehabilitation and follow-up visit optimization is achieved, and the accuracy and timeliness of voice function protection and rehabilitation are improved.

The application relates to the technical field of smart home devices, and provides a voice function optimization method and device for a smart home device, which comprises the following steps: determining human body posture attribute information in at least one home layout drawing, setting attribute information of a smart home device, and acquiring voice data corresponding to the human body posture attribute information collected by the smart home device under the setting attribute information; determining a voice interaction success rate under the condition of the human body posture attribute information in the home layout drawing and the setting attribute information of the smart home device based on the voice data; and performing optimization processing on the smart home device based on the voice interaction success rate. The method provided by the application can effectively improve the efficiency of voice interaction optimization processing of the smart home device.

The invention provides a message processing method and device, electronic equipment and a storage medium, and belongs to the technical field of artificial intelligence. The method comprises the following steps: displaying an interaction page between a first user account and a second user account; in response to the obtained new multimedia message, displaying the multimedia message on the interactive page; and under the condition that the multimedia message is in a non-voice form, displaying a voice synthesis result of the multimedia message, and broadcasting the voice synthesis result based on the voice characteristics of the target role. For a private message scene, the multimedia message processing method and the multimedia message processing device support converting multimedia messages in a non-voice form into voice and broadcasting the voice according to voice features of a specific role. The voice conversion function enriches man-machine interaction modes in a private message scene, and meets diversified conversion requirements of users. In addition, based on the voice conversion function, the message content sent by the opposite side can be quickly and conveniently obtained, and thus the communication efficiency can be improved.

1. The name of the design product: intelligent interactive device (research terminal). 2. The use of the design product: intelligent interactive device for controlling research interaction with users and having text-to-speech function. 3. The design points of the design product: in shape. 4. The picture or photo that best indicates the design points: perspective view 1.

The invention relates to the technical field of household appliances, and discloses a heating device which comprises a shell, a voice module and an integrated support. The voice module is arranged in the shell and comprises a loudspeaker and a microphone; the integrated bracket is fixed in the shell; the integrated support is provided with a first holding part and a second holding part, the loudspeaker is fixed to the first holding part, the microphone is fixed to the second holding part, the single integrated support is introduced, the loudspeaker and the microphone are fixed through the first holding part and the second holding part on the integrated support respectively, and integration and modularization of the voice module are achieved; the voice module can be assembled only by adding one integrated support, integration and assembly of the voice module can be completed with least newly added parts, other original structural components of equipment do not need to be changed at all, the integrity of the original design is reserved to the maximum extent, and the cost is reduced. The aim of adding the voice function on the heating device is achieved, and meanwhile assembling simplification and structure compatibility are taken into consideration.

The invention relates to the technical field of computer software, and particularly discloses a brain voice evaluation method and system based on a DSM-V cognitive domain, and the method comprises the steps: obtaining the voice information of a user; feature extraction is carried out on the voice information based on a DSM-V cognitive domain to obtain spoken language feature information, and the spoken language feature information comprises vocabulary reaction feature information, vocabulary semantic feature information and vocabulary association feature information; according to the vocabulary reaction feature information, obtaining multiple stages of vocabulary feature information, and according to all the stages of vocabulary feature information, obtaining a vocabulary reaction feature evaluation value; and obtaining feature information of a plurality of vocabulary nodes according to the vocabulary semantic feature information. According to the method, a foundation is laid for comprehensive assessment by obtaining the voice information, multi-dimensional spoken language feature information is extracted based on the DSM-V cognitive domain, the brain voice function can be deeply analyzed, the problem that traditional assessment dimensions are single is solved, and for example, the root of the problem can be accurately positioned in language disorder diagnosis.

To solve the problems related to resume writing and interviewing practice that foreign workers face when seeking employment in Japan.SOLUTION: Provided is a system comprising: means that when a foreign worker seeks employment in Japan, makes ChatGPT learn resume patterns so as to convert the content of a resume in which the person whose first language is not Japanese enters in English or his / her native language into a natural Japanese-style resume; and means for providing a bot combined with a voice function, which can conduct interviewing practice tailored to various industries in interviewing practice.SELECTED DRAWING: Figure 1

The invention provides a noise filtering method and device, a computer readable storage medium and electronic equipment. The method comprises the following steps: in a first preset environment, noise data of operation of the electrical equipment in each function mode is obtained, first noise data corresponding to each function mode is obtained, the first preset environment represents an environment without other sounds except the electrical equipment in a preset closed space before the electrical equipment leaves a factory, and the first noise data is sent to the electrical equipment; the first noise data comprises operation noise of the electrical equipment; in a second preset environment, noise data of operation of the electrical equipment in each function mode is obtained, and second noise data corresponding to each function mode is obtained; and filtering the first noise data and the second noise data to obtain target voice data. According to the invention, the problem of poor user experience caused by the fact that the voice function of the electrical equipment is greatly influenced by noise in the prior art is solved.

Systems and methods for temporarily associating user accounts with voice-enabled devices are disclosed. A voice-enabled device, which may be associated with a default account, may be situated in an environment, such as a hotel room. A user may desire to utilize the voice-enabled device to perform one or more actions. In examples, some or all of the actions may require the use of a system resource not available to the default account. In these examples, the user may provide contact information to be used to send a message to personal device of the user to acquire user account information. Upon receipt of such information, the user account may be temporarily associated with the voice-enabled device, and may be dissociated from the voice-enabled device upon the lapse of a period of time and / or occurrence of a dissociation event.

The utility model relates to a voice function detection system and device.The voice function detection system comprises a voice module and a filtering and rectifying circuit, and the filtering and rectifying circuit is electrically connected with the voice module and used for receiving voice signals transmitted by the voice module, conducting filtering and rectifying on the voice signals to obtain input electric signals and sending the input electric signals to the voice module; the input electric signal is transmitted to the voice module; and the voice module is used for receiving the input electric signal and judging whether the voice input and output functions of the voice module can operate normally or not according to the input electric signal.

The invention relates to the technical field of vehicle-mounted system testing, in particular to a testing method for simulating a manual voice testing function, which comprises the following steps of: by simulating a real driving condition, placing a simulation cockpit carrying a vehicle-mounted system in a simulated environment; the defects that a traditional manual test method is time-consuming and labor-consuming, and multiple test conditions are difficult to copy are avoided; an artificial mouth is used as a reference source to simulate real voice output, so that the consistency and repeatability of the test are ensured, and the accuracy of the test is improved; by means of simulating real driving conditions, simulating voice output by using an artificial mouth, installing a measuring microphone and a hand-free microphone, performing automatic analysis and anomaly detection, generating a detailed test report and the like, the problems that in the prior art, the test of a vehicle-mounted voice interaction system mainly depends on manpower, time and labor are consumed, and the test efficiency is high are solved. And a plurality of voice scenes and environment noise conditions are difficult to fully cover.

The utility model discloses a cultural and creative entity ticket which comprises a token main body, one side surface of the token main body is provided with a groove, the groove is buckled with a cover plate, the interior of the groove is divided into two placing grooves through a partition plate, and one placing groove is provided with an electronic ticket; the other placing groove is provided with a power supply module, a radio frequency tag identification module, a control processing module and a voice broadcast module, and the power supply module supplies power to the radio frequency tag identification module, the control processing module and the voice broadcast module; a sinking groove is formed in the other side surface of the token main body; and a scenic spot route map or a scenic spot marker embossment is arranged in the sinking groove. The electronic ticket and the scenic spot explanation voice function are integrated on the token, in addition, the token body is further provided with the scenic spot route map or the scenic spot marker embossment, the token can be tied on the waist or held by a hand, the token is different from an existing paper ticket, the touring experience feeling, the use feeling and the emotional value of tourists are improved, and the tokens can be collected for dwell.

The invention relates to the technical field of power production safety management, and discloses an intelligent management and control system suitable for power field operation safety. The theoretical basis of the method is that safety operation = creation and maintenance of safety and operation conditions + zero error of people. Comprising a field operation risk analysis module, a safety reminding and checking module, a positioning tracking module, a monitoring analysis module, a training education module and a data management module. A function of automatically acquiring work ticket and work ticket information by associating the work ticket and the work ticket; through operation risk analysis, dangerous point identification and management and control measure functions are realized; safety reminding of an operator is realized through a voice function; an intelligent audio and video system is utilized to monitor an operation site in an omnibearing and dead-corner-free manner, unsafe behaviors of people, unsafe states of equipment and unsafe factors of the environment are accurately recognized, and an audible and visual alarm is given out in time; and a self-learning function is integrated with historical records to continuously perfect an operation dangerous point identification management and control database, and safety training data is perfected.

The application discloses a voice processing method, comprising: receiving wake-up sound area information of a user who wakes up a vehicle voice function in a vehicle cabin; determining an initial rejection mode of each sound area in the vehicle multi-sound area cabin according to the wake-up sound area information; receiving a user voice request forwarded by the vehicle after the vehicle voice function is woken up and dialogue sound area information confirmed according to the user voice request; and updating the rejection mode of the corresponding sound area according to the user voice request and the dialogue sound area information to determine the rejection mode of each sound area. In the application, after receiving the voice request, the rejection mode of each sound area corresponding to the voice request is confirmed according to the voice request and the voice request, so that the rejection demand of the multi-sound area voice interaction in the vehicle cabin can be met. Meanwhile, the rejection mode of each sound area is updated along with the voice interaction, so that in the multi-sound area interaction scene, the voice request rejection accuracy is high, and the user experience is better.

To provide a system and a method for monitoring and optimizing a user interaction session in an interactive voice response system during a human-computer interaction.SOLUTION: A user interaction management system (100) for monitoring and optimizing a user interaction session includes a conversation controller module (109). The conversation controller module (109) receives a voice feature during the user interaction session, processes output of ASR (automatic speech recognition) and NLU (natural language understanding) and optimizes duration time of the user interaction session. A conversation controller may suggest a TTS (text to speech) module (107) to increase or decrease the rate of speech.SELECTED DRAWING: Figure 1A

A method of controlling a battery-powered remote controller to decrease a duty cycle to allow continued operations despite the quantity of the battery is bad. The method determines a drop in voltage of the battery in standby mode as voltage of the battery is being read. When receiving a command to activate a voice function, determining whether the drop in voltage in the standby mode is greater than or equal to a preset value. If the drop is greater than or equal to the preset value, the method then determines whether the drop in voltage falls in a preset range. If the drop falls in the preset range, the method regulates a duty cycle of the pulse signal activating the voice function, and activates the voice function as required. A remote controller and a non-transitory storage medium are also provided.

1. Name of the product in this design: Instant Messaging Device. 2. Purpose of this design: It is intended for use as a portable instant messaging device in complex environments, and features video and voice functions. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key points: 3D view 1.

This wearable device may provide torque to a user who is exercising, obtain an indicator about the extent of burden on the body of the user on the basis of at least one of exercise information about exercise performance of the user, biometric information of the user, or setting information set so that the wearable device provides the torque, select a sentence to be used for voice generation of the wearable device on the basis of the indicator, generate a voice on the basis of the selected sentence, and provide the generated voice to the user.

To provide systems, computer-implemented methods and non-transitory computer-readable media for selecting features of speech to be used to build or train a mathematical model for detecting or diagnosing a medical condition.SOLUTION: A computing device improves the performance of a mathematical model by appropriately selecting features 1021, 1022 to be used with the mathematical model. The device computes a feature selection score 1031 for each acoustic feature 1021 and each language feature 1022, and then uses the scores 1031 to select the features 1021, 1022 with the highest scores 1031. The device then uses the selected features 1021, 1022 to train and deploy a mathematical model. In some implementations, the device may use computed prompt selection scores 1041 to select prompts, and use the deployed mathematical model with the selected prompts.SELECTED DRAWING: Figure 10

This application provides a cell handover method, apparatus, network device, and storage medium, relating to the field of communication technology. It solves the technical problem that a terminal may be unable to conduct voice calls based on the visited network, leading to voice call interruption and reducing the effectiveness of voice services. The method includes: if the network quality of a second cell meets preset conditions, determining whether the terminal has voice functionality; if the terminal has voice functionality but does not have the preset voice functionality, determining whether the network quality of a third cell meets the preset conditions, wherein the third cell is a cell in the neighboring cell set other than the second cell, and the terminal's home network includes the third cell; if the network quality of the third cell meets the preset conditions, handing over the terminal's serving cell to the third cell.

The invention discloses a network security system driven by artificial intelligence. The network security system is designed to detect, analyze and relieve wide network threats in real time. The network security system comprises a data receiving module which is responsible for gathering diversified data types, including images from various sources, text content and URLs (Uniform Resource Locator). The AI analysis module processes the received data using image recognition, optical character recognition, and semantic analysis, and identifies potential threats, such as phishing attacks, fraud, and propagation of malicious content. And a link analysis module evaluates the security of the URL through the dynamic white list and black list, and performs AI-based evaluation on unknown links. The audio interface expands threat detection to audio-based communication, integrates voice-to-text and text-to-voice functions, and enables the system to analyze fraud or malicious intentions that may exist in a telephone call.