111 results about "Teleconference" patented technology

This document relates to providing adaptive teleconferencing experiences using generative image models. For example, the disclosed implementations can employ inpainting and / or image-to-image restyling modes of a generative image model to generate images for a teleconference. The images can be generated based on prompts relating to the teleconference. Users can be superimposed on the generated images, thus giving the appearance that the users are present in an environment generated by the generative image model.

A method for sentiment based enhancement of investor relations communication. The method includes: receiving an enhancement request including an original current earnings call transcript; processing, of the original current earnings call transcript, a plurality of original sentiment-pertinent transcript sentences using language models to respectively obtain a plurality of transcript sentence ranks; identifying a subset of the plurality of transcript sentence ranks each indicating a sentiment improvement; producing, from the original current earnings call transcript, a new current earnings call transcript including a set of enhanced sentiment-pertinent transcript sentences respectively mapped to the subset of the plurality of transcript sentence ranks; and providing the new current earnings call transcript in response to the enhancement request.

One or more systems, devices, computer program products and / or computer-implemented methods of use provided herein relate to teleconferencing and, more specifically, to the gathering and conveying of information over a teleconference by a digital delegate. A computer-implemented system can comprise a memory that can store computer executable components. The computer-implemented system can further comprise a processor that can execute the computer executable components stored in the memory, wherein the computer executable components can comprise a digital delegate component that accesses a teleconference via a computer application as a teleconference participant and an analysis component that analyzes teleconference monitoring information of the teleconference to determine responses to one or more defined points of interest determined by a meeting invitee.

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).

Voice anonymization systems and methods are provided. Voice anonymization is done on the speaker's computing device and can prevent voice theft. The voice anonymization systems and methods are lightweight and run efficiently in real time on a computing device, allowing for speaker anonymity without diminishing system performance during a teleconference or VoIP meeting. The anonymization system outputs a transformed speaker voice. The anonymization system can also generate a voice embedding that can be used to reconstruct the original speaker voice. The voice embedding can be encrypted and transmitted to another device. Sometimes, the voice embedding is not transmitted and the listener receives the anonymized voice. Systems and methods are provided for the detection of voice transformations in received audio. Thus, a listener can be informed whether the speaker voice output from the listener's computing device is the original speaker's voice or a transformed version of the original speaker voice.

A method includes, during a teleconference between a first audio input / output device and a second audio input / output device, receiving, at an analysis and response device, a signal indicating a spoken command, the spoken command associated with a command mode. The method further includes, in response to receiving the signal, generating, at the device, a reply message based on the spoken command, the reply message to be output to one or more devices selected based on the command mode. The one or more devices includes the first audio input / output device, the second audio input / output device, or a combination thereof.

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 back-end server system responds to a request for web-based browser or dedicated application interaction associated with an entity with which the user holds at least one account; performs basic authentication to grant a basic level of remote user access over the Internet to the user device; performs elevated authentication and grants access (i) to an elevated level of remote user access to the at least one account and (ii) access to a pre-authenticated teleconference session.

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.

The telephone relay device (30) is communicably connected to a teleconference server that relays an audio call between communication terminals participating in a teleconference via an IP network. The telephone relay device includes: a participation request unit (32) that requests the teleconference server to permit participation in the teleconference in response to receiving, from the teleconference server, an audio quality reduction notification indicating that audio quality of the audio call between the communication terminals becomes less than a predetermined level; a call making unit (33) that makes a call to a telephone terminal associated with the communication terminal in response to being permitted to participate in the teleconference; and a call relay unit (34) that relays an audio call of the teleconference between the telephone terminal and the teleconference server in response to receiving a response to the call from the telephone terminal.

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.

The present disclosure relates to a video teleconferencing method between at least two remote video teleconferencing apparatuses in communication via a communication network, wherein each video teleconferencing apparatus comprising a camera and a display, wherein at least one video teleconferencing apparatus, called first video teleconferencing apparatus, comprising a camera, called first camera, whose field of view covers at least partially a viewpoint of an end-user of the first video teleconferencing apparatus and a display, called first display, wherein the method comprises the following steps: -capturing (100), by the first camera, a first video representative of a self-view of the end-user of the first video teleconferencing apparatus on a reflective surface of a mirror located in front of the end-user and in the field of view of the first camera; -extracting (110), from the first video, at least one pixel region surrounding the self-view of the end-user; and -transmitting (120) first video data representative of said at least one extracted pixel region to at least one remote video teleconferencing apparatus.

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).

1. Name of the Design Product: Conference TV Terminal (Remote Collaboration). 2. Use of the Design Product: For video conferencing, telephone conferencing, and remote collaboration. 3. Design Key Points of the Design Product: Lies in the overall shape of the product. 4. Picture or Photograph that Best Illustrates the Design Key Points: Perspective view.

Participants may use one or more devices for engaging in a meeting, such as phones, conferencing devices, and / or computers. The devices include microphones that capture speech for determining the presence of distinct participants. Speech signals originating from different participants, or microphones, may be determined and associated with the participants. For example, microphones may be directional and more sensitive to sound coming from one or more specific directions than sound coming from other directions. By associating an individual with a microphone, or set of microphones, overlapping voices may be disambiguated to provide clear voice streams that aid in producing a clear transcript indicating the speech of the participants, respectively. An identity of the participants may be determined using voiceprint and / or voice recognition techniques.

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.

The embodiment of the invention discloses a remote conference management method and device, equipment and a medium. The method comprises the following steps: acquiring audio data of each conference member in a teleconference; wherein the conference members comprise a first conference member and at least one second conference member; integrating the audio data of the first conference member and the at least one second conference member to obtain an audio data set; and respectively transmitting the audio data set to equipment corresponding to the first conference member and the at least one second conference member, and playing corresponding audio data based on the audio data set and a set playing rule. According to the technical scheme, the corresponding audio data is played according to different conference members and the set playing rule, so that the flexibility and convenience of the teleconference can be improved, and diversified requirements of different conference stages and discussion topics can be met.

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

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.

This document describes techniques, methods, systems, and other mechanisms for improving interactions between insurance agents, insurance providers, and insurance customers while ensuring that such interactions comply with regulatory requirements. This includes providing unique user interfaces that facilitate efficient communication between computing devices operated by insurance agents and computing devices operated by insurance customers to ensure that electronic scope of appointment forms are provided to potential customers and executed in accordance with government regulations prior to face to face or telephonic meetings between an insurance agent and an insurance customer.