77 results about "Teleconference" patented technology

Disclosed in the present invention is a Gaussian mixture shape method applicable to dynamic modeling of the head of a human body. A Gaussian mixture shape is composed of a neutral Gaussian model representing a neutral expression and a plurality of basic expression Gaussian models corresponding to a plurality of basic expressions. Each Gaussian model comprises a group of three-dimensional Gaussian functions having information such as position, shape, and color. A human face and head model of any expression is efficiently generated by linearly mixing the neutral Gaussian model and the basic expression Gaussian models. The method comprises training image acquisition and preprocessing, Gaussian mixture shape training, and human head animation generation. The present invention has higher drawing quality, can retain more high-frequency details of the head of the human body, and improves the sense of reality and fidelity of drawing; and the present invention has higher operating speed which is more than five times higher than that of the current most advanced method. Therefore, the present invention can be widely applied to movies, animation production, online games, teleconferences, etc., and has high practical value.

Techniques for transcribing and / or summarizing multimedia collaboration sessions are discussed herein. For example, users can communicate within a teleconferencing meeting associated with a channel. In some examples, a first machine learning model may be configured to receive audio-visual data and user interaction data (e.g., selected emojis, detected gestures, messages or text input by a user, a thread of messages, etc.) and output a teleconferencing meeting summary. In some examples, the teleconferencing meeting summary can include one or more highlights, action items, and / or artificial intelligence (AI) notes generated using a machine learning model(s).

The invention discloses a multifunctional earphone storage box and a dual-mode audio system, and belongs to the technical field of intelligent software and hardware equipment.The multifunctional earphone storage box comprises a device product, an array microphone, a built-in banded loudspeaker, an earphone cabin cover, a TWS wireless Bluetooth earphone, a Type-c usb charging port, a liquid crystal display panel and a multifunctional button. According to the mobile phone, the text conversion and translation service is supported, the translated and translated text content can be edited and modified, a user can conveniently manage and utilize recording data, the mobile phone APP supports voiceprint information management, a speaker can be created, modified and deleted, and the mobile phone APP is convenient to use, high in practicability and high in practicability. And meanwhile, a large model conference summary capability is requested from a server, summary summarization and to-be-processed extraction are realized, conference efficiency is improved, the system can be used as a mobile phone loudspeaker and an external microphone after Bluetooth audio connection, and the system is suitable for a remote conference scene and is not limited by third-party conference software or a teleconference.

A method of setting up a teleconferencing system using mobile devices is disclosed. Participants with mobile devices in multiple rooms may communicate based on the disclosed teleconferencing application executed on the mobile devices. Each room has one of the mobile devices designated as a leader device for that room. The leader devices may communicate using VoIP over the Internet. Other mobile devices in a room are designated as client devices associated with the leader devices. Participants operating the client devices can talk to other participants in that room and the other rooms through the client device communicating through the leader device of the room. The leader device synchronizes the playing of received audio between all of the client devices in the room. The leader device includes a selection algorithm that selects one of the devices to act as the microphone for audio signals generated in the room.

The invention relates to the technical field of intelligent contracts, and discloses an intelligent contract safety generation and risk insight method fusing a large language model and a digital signature, and the method comprises the steps: firstly obtaining a telephone conference voice, carrying out the telephone band processing, and carrying out the system sound interference suppression, and obtaining a purified signal; positioning a short near-end fragment based on the purification signal, extracting an energy peak moment, and setting an early window and a late window; estimating a fundamental frequency in the early window and constructing a harmonic set, calculating a normalized double-coherence intensity set average value of the early window and the late window, constructing a characteristic index and forming a time-varying gating weight field through monotone mapping; obtaining a word-level timestamp through automatic speech recognition, and calculating word availability in combination with a gating weight field; extracting an offer of the speaker A, collecting response words of the speaker B in a preset time window, and calculating response reliability according to word availability and time closeness; and generating a clause draft, judging whether the clause draft is included in the first draft of the contract, and outputting risk clauses of the contract draft.

Described herein are methods, systems, devices, and processor readable storage devices for use during a teleconference among a plurality of users that each uses a respective device, of a plurality of devices, to participate in the teleconference. Certain such embodiments use positional information to cause displaying, by a device that is provided with the positional information, of a respective graphical indicator that indicates where a region of interest is within or relative to the viewport that is being displayed on the device.

A controller of a virtual conferencing system commands an online conference platform to create an online main conference room associated with a meeting, and commands a telephonic conference platform to create a telephonic main conference room associated with the meeting. The controller further commands at least one of the online and telephonic conference platforms to join the online main conference room and the telephonic main conference room together so audio is passed therebetween to form an integrated main conference room. A similar process is used to create an integrated screener room. The controller stores a question queue identifying both online and telephonic attendees who have questions. Attendees who have an unscreened question are moved into either an online or telephonic part of the integrated screener room depending on the way the attendee joined the meeting. Moderators move both telephonic and online attendees with screened questions into the main conference room.

Systems and methods are provided for identifying polling questions from a conference call discussion. One or more text strings are identified, the one or more text strings comprising a textual form of one or more spoken phrases provided by one or more participants of a conference call. The one or more text strings are provided as input to a trained machine learning model. One or more outputs are obtained from the trained machine learning model. A spoken phrase of the one or more spoken phrases provided by the one or more participants is extracted from the one or more outputs that comprises a confidence level that a question associated with a poll during the conference call. In response to a determination that the confidence level satisfies a confidence criterion, the spoken phrase is designated as a polling question presented during the conference call.

A method (300, 400) of operating a telephony system (100), said system comprising a: client server (170), configured to: access a conferencing server (140) for establishing a conference call; connect a plurality of terminals (180) to the conferencing server; and communicate with a UE (160) so as to receive a call from the UE; the method comprising the steps of, by the client server: detecting initiation of a call between the UE and a given one of the plurality of terminals (320, 420); identifying the UE and said given terminal; assessing whether the detected call is a conference call or a non-conference call; in response to assessing that the call is a non-conference call: requesting, from the conferencing server, establishment of a conference call for the identified UE and the given terminal and corresponding to said non-conference call (330-2, 430); receiving, from the conferencing server, access information for accessing the requested conference call (340, 440); and distributing the access information to each of the plurality of terminals (360, 460), thereby permitting any of said terminals to access the conference call.

A method of operating a telephony system, the system including a client server to access a conferencing server for establishing a conference call, connect a plurality of terminals to the conferencing server; and communicate with a UE so as to receive a call from the UE;, the method including, by the client server, detecting initiation of a call between the UE and a given one of the plurality of terminals; identifying the UE and the given terminal; assessing whether the detected call is a conference call or a non-conference call; in response to assessing that the call is a non-conference call: requesting, from the conferencing server, establishment of a conference call for the identified UE and the given terminal and corresponding to the non-conference call; receiving, from the conferencing server, access information for accessing the requested conference call; and distributing the access information to each of the plurality of terminals, thereby permitting any of the terminals to access the conference call.

A method and apparatus for signaling multiple audio mixing gains using a single Real-Time Transport Protocol (RTP) header extension in a teleconference is provided. The method includes: receiving an input audio stream from a 360-degree video / audio stream, the input audio stream including multiple mixing gains from the input audio stream and an overlay audio stream; declaring a single RTP header extension for all mixing gains (including one or more of an element identifier, a length of the extension element, and a magnitude of the mixing gain); and signaling the mixing gains using the single RTP header extension. The single RTP header extension can use a one-byte header extension format or a two-byte header extension format, and the one-byte header extension format or the two-byte header extension format is declared using the Session Description Protocol (SDP).

This document relates to distributed teleconferencing. Some implementations can employ personalized enhancement models to enhance microphone signals for participants in a call. Further implementations can perform proximity-based mixing, where microphone signals received from devices in a particular room can be omitted from playback signals transmitted to other devices in the same room. These techniques can allow enhanced call quality for teleconferencing sessions where co-located users can employ their own devices to participate in a call with other users.

The invention discloses a voice separation and paragraph affiliation method and system for a teleconference scene, and relates to the technical field of voice recognition. The method comprises the following steps: a conference platform triggers a voice record of a target conference window to generate a conference audio; according to the first engine, executing two-way voice separation processing on the conference audio, and constructing a voice logic map; according to a second engine, matching a target conference template, executing voice fragment attribution based on automatic conference record typesetting according to the voice logic map, and generating a target conference record; and carrying out platform database storage on the target conference record, and carrying out calling management by permission setting. The technical problems that in the prior art, due to the fact that voice interweaving in a multi-party conference is difficult to effectively distinguish spokesmen and speaking content logic structures of the spokesmen, conference record affiliation is not clear, and editing efficiency is low are solved, two-way semantic logic separation and structured affiliation of the voice content of the teleconference are achieved, and user experience is improved. And the conference recording accuracy and the automatic typesetting efficiency are improved.

A participant computing device connected to a teleconferencing session receives mixed audio data from a teleconference computing system orchestrating the session. The mixed audio data comprises a mix of audio captured by other devices connected to the session. The mixed audio and audio captured by the other devices prior to receipt of the mixed audio data are compared to determine that the participant computing device is co-located with at least one of the other devices. Information indicating that some of the other devices are featured devices that most actively contribute to the mixed audio is obtained. Co-location information for each of the featured devices is received. The co-location information is indicative of whether a featured device is co-located with the participant computing device. Based on the co-location information, a determination is made that one of the featured devices is co-located with the participant computing device.

A controller of a virtual conferencing system commands an online conference platform to create an online main conference room associated with a meeting, and commands a telephonic conference platform to create a telephonic main conference room associated with the meeting. The controller further commands at least one of the online and telephonic conference platforms to join the online main conference room and the telephonic main conference room together so audio is passed therebetween to form an integrated main conference room. A similar process is used to create an integrated screener room. The controller stores a question queue identifying both online and telephonic attendees who have questions. Attendees who have an unscreened question are moved into either an online or telephonic part of the integrated screener room depending on the way the attendee joined the meeting. Moderators move both telephonic and online attendees with screened questions into the main conference room.

A telephone conferencing system capable of conducting telephone conferences smoothly is provided. According to the invention, a speech determination unit (2) determines whether the speech of each participant in a telephone conference is a statement or an interruption. A speech output control unit (4) performs control such that the speeches of the multiple participants are output to the communication terminals of the multiple participants respectively. When another participant speaks while one of the multiple participants is speaking, the speech output control unit (4) performs control to suppress the output of the other participant's speech. A counting unit (6) counts the number of speech conflicts for each participant. A count display control unit (8) performs control such that a display related to the count is displayed on the communication terminals of the multiple participants.

The utility model relates to the technical field of teleconferences, and provides a teleconference system, which comprises a rotating support, an acquisition support is fixedly mounted in the rotating support, an acquisition camera for face recognition and information acquisition is fixedly mounted in the acquisition support, and a camera is fixedly mounted in the acquisition camera. A laser distance measuring sensor for detecting the position of the acquisition camera is arranged on one side of the acquisition camera; the utility model has the advantages that the rotary support and the acquisition support which can be rotationally adjusted are arranged to support and adjust the camera for face recognition and image acquisition, so that the working direction of the camera can be conveniently adjusted according to requirements, and the main body display content of a conference can be switched; according to the remote conference system, the cameras are arranged, the laser distance measuring sensor is arranged on one side of each camera to detect the distance between the cameras, the collecting support is controlled according to detected data to conduct local fine adjustment on the positions of the cameras, the cameras can work more clearly, and then the use convenience of the remote conference system is effectively improved.

The application relates to the technical field of intelligent contract, and discloses an intelligent contract security generation and risk insight method fusing a large language model and a digital signature, which comprises the following steps: first, acquiring a telephone conference voice and performing telephone voice band processing and system sound interference suppression to obtain a purified signal; positioning a short near-end segment based on the purified signal, extracting an energy peak time and setting an early window and a late window; estimating a fundamental frequency in the early window and constructing a harmonic set, calculating a normalized bicoherence intensity set average value of the early window and the late window, constructing a characteristic index and forming a time-varying gating weight field through a monotone mapping; then, acquiring a word-level timestamp through automatic speech recognition, and calculating word availability in combination with the gating weight field; extracting an offer of a speaker A, collecting response words of a speaker B in a preset time window, and calculating response reliability according to the word availability and time closeness; generating a clause draft according to the response reliability, judging whether the clause draft is included in a contract preliminary draft, and outputting a risk clause of the contract draft.

A communication terminal, which is mounted on a mobile body and is used for performing a teleconference, is provided with: a reception unit that receives a plurality of users in the mobile body as participants of the teleconference; a speaker specifying unit that specifies a speaker from among a plurality of users accepted as participants of the teleconference by the accepting unit; and a control unit that performs control for highlighting the user identified as the speaker by the speaker identification unit on the teleconference system.

This invention discloses a method and system for speech separation and segment attribution in remote conferencing scenarios, relating to the field of speech recognition technology. The method includes: triggering speech recording in a target meeting window via a conferencing platform to generate meeting audio; performing two-way speech separation processing on the meeting audio using a first engine to construct a speech logic graph; matching a target meeting template using a second engine and performing speech segment attribution based on automated meeting record layout according to the speech logic graph to generate a target meeting record; and storing the target meeting record in a platform database for retrieval and management using access settings. This invention solves the technical problem in existing multi-party conferencing systems where speech interleaving makes it difficult to effectively distinguish speakers and the logical structure of their statements, leading to unclear meeting record attribution and low editing efficiency. It achieves two-way semantic logic separation and structured attribution of remote meeting speech content, improving the accuracy of meeting records and the efficiency of automatic layout.

Solutions are disclosed that provide automated audience response estimation (sentiment analysis) and presenter feedback. Examples capture a plurality of multi-modal signals from a first multi-participant interaction session, such as capturing an audio feed, a video feed, a chat, and actions (e.g., hand-raising) from a video teleconference. Timing information is correlated, enabling accurate sentiment analysis across the multi-modal signals, such as nodding in agreement, detected in the video feed, is correlated with spoken words, captured in the audio clip and identified in an automated transcript. This enables reporting audience sentiment to the presenter, in near-real-time (i.e., during the teleconference) in some examples. Some examples combine multi-modal sentiment analysis results from multiple teleconferences in order to create or train a presentation coach that is able to suggest improvements to planned presentations. Some examples are able to identify a particular audience member (e.g., a VIP), and perform individualized sentiment analysis for that person.

A telephone relay device (30) is connected to a teleconference server to allow communication with the teleconference server, which relays audio calls between participating communication terminals via an IP network. The telephone relay device includes a participation request unit (32), a call execution unit (33), and a call relay unit (34). In response to receiving an audio quality degradation notification from the teleconference server indicating that the audio quality of the audio call between the communication terminals is below a predetermined level, the participation request unit (32) requests permission from the teleconference server to participate in the teleconference. In response to permission to participate in the teleconference, the call execution unit (33) makes a call to the telephone terminal associated with the communication terminal. In response to receiving a response to the call from the telephone terminal, the call relay unit (34) relays the teleconference audio call between the telephone terminal and the teleconference server.

The invention provides a video conference method comprising the following steps: obtaining a first video stream collected by an image collection device in a first state, the first video stream comprising image information of a target participant; outputting the first video stream to the teleconference device; in response to the detected target behavior which is executed by the target participant and associated with the target object, the image acquisition device is adjusted from the first state to the second state, and the image acquisition parameters of the image acquisition device are different in the first state and the second state; obtaining a second video stream collected by the image collection device in a second state, wherein the second video stream comprises image information of the target object; and outputting the second video stream to the remote conference equipment. The invention further provides a video conference device, equipment, a storage medium and a program product.

The invention discloses a remote projection screen terminal control system and method of a remote interactive interface, and relates to the technical field of electric digital data processing. The remote screen projection terminal control system of the remote interactive interface comprises an input response judgment module, a terminal response control module and a response return judgment module. According to the method, whether the remote projection screen terminal response is triggered or not is judged by combining the input interaction signal interference influence result, if the remote projection screen terminal response is not triggered, input interaction signal transmission optimization adjustment is prompted to be executed, and otherwise, the image quality of the remote projection screen terminal in the remote projection screen terminal response process is evaluated; and whether the remote screen projection terminal is triggered to respond to return or not is judged, so that terminal picture quality control in the screen projection interaction process of the teleconference is more efficiently carried out, and the problem that interference differences between the terminal picture quality and interaction signals are not sufficiently associated in the screen projection interaction process of the teleconference in the prior art is solved.

Using a camera pointing downward, a working area below eye level can be displayed by a headwear for users, so that the user can maintain healthy sitting or standing posture while working on patients or objects located below eye level. Using two or more cameras pointing downward, three-dimensional views with accurate and natural depth perception of a working area can be displayed by a headwear for users. Additional functions including eye protection, zoom-in, zoom-out, on-off, lighting control, overlapping, and teleconference capabilities are also supported using electronic, video and audio devices associated with the headwear. The headwear can also comprise a face shield designed to protect the user from hazardous droplets, aerosols, harmful wavelengths of light, heat, sparks, flash burn, debris and / or flying objects. Contactless control devices can be used to control the functions of the headwear.

Aspects of the present application provide an audio mixing method for a teleconference and a computer readable medium, the method comprising: receiving, by a processing circuit of a first device, a first media stream and a second media stream of the teleconference from a second device, where the first media stream comprises a first audio and the second media stream comprises a second audio; receiving default weight information from the second device, where the default weight information indicates a first audio weight for weighting the first audio and a second audio weight for weighting the second audio; determining a first audio weight for weighting the first audio and a second audio weight for weighting the second audio based on the default weight information; and generating, by processing circuitry of the first device, a mixed audio by combining the weighted first audio based on the first audio weight and the weighted second audio based on the second audio weight.

A computer-implemented method of accessing a user account from a video terminal system. The video terminal system is used for carrying out a video telephone conference and comprises a main processor, a control interface and video conference hardware. The computer-implemented method comprises: obtaining an access token configured to grant access rights to user account information associated with a user account to a video terminal system; and retrieving the user account information using the access token.

This document relates to adaptive presentation of content during a teleconference. For instance, the disclosed techniques can employ one or more attention signals to infer that a user's attention is directed to certain content presented during a particular time period during a teleconference. Then, other content that the user may have missed can be summarized using a generative machine learning model. A summary of the potentially-missed content can be output to the user during the teleconference. In this manner, users who may miss presentation of certain content can remain informed during the course of the teleconference.