Conference call system, method, and computer program product

A conference call system comprises an input interface for receiving during a conference call at least two input streams of audio signal, each from another source. A selection unit is connected to the input interface, for selecting a number of dominant speaker streams out of the input streams, the number being less than or equal to a maximum number of dominant speakers value and each of the dominant speaker streams representing speech from a respective dominant speaker. A mixer is connected to the selection unit, for mixing the selected streams into an output stream. The conference call system comprises an output interface for outputting the output stream and a selection control unit connected to the selection unit and the input interface, for dynamically setting, during the conference call, the maximum number of dominant speakers value based on dynamics of the conference call.

Multi-channel telephony

Techniques for providing multi-channel telephony are described herein. In different aspects, a multi-channel telephony apparatus may include a sound interface configured to process a multi-channel sound signal including at least one of causing speakers to emit multi-channel sound or receiving multi-channel sound from microphones and a network interface configured to transmit the multi-channel sound signal from the sound interface to one or more communication lines of a telephone network. In some aspects, a dialer may be configured with the network interface to establish a connection with the telephone network.

Active speaker identification

Procedures for identifying clients in an audio event are described. In an example, a media server may order clients providing audio based on the input level. An identifier may be associated with the client for identifying the client providing input within the event. The ordered clients may be included in a list which may be inserted into a packet header carrying the audio content.

Conference call system, method, and computer program product

A conference call system comprises an input interface for receiving during a conference call at least two input streams of audio signal, each from another source. A selection unit is connected to the input interface, for selecting a number of dominant speaker streams out of the input streams, the number being less than or equal to a maximum number of dominant speakers value and each of the dominant speaker streams representing speech from a respective dominant speaker. A mixer is connected to the selection unit, for mixing the selected streams into an output stream. The conference call system comprises an output interface for outputting the output stream and a selection control unit connected to the selection unit and the input interface, for dynamically setting, during the conference call, the maximum number of dominant speakers value based on dynamics of the conference call.

Systems and methods for filtering unwanted sounds from a conference call using voice synthesis

To filter unwanted sounds from a conference call, a first voice signal is captured by a first device during a conference call and converted into corresponding text, which is then analyzed to determine that a first portion of the text was spoken by a first user and a second portion of the text was spoken by a second user. If the first user is relevant to the conference call while the second user is not, the first voice signal is prevented from being transmitted into the conference call, the first portion of text is converted into a second voice signal using a voice profile of the first user to synthesize the voice of the first user, and the second voice signal is then transmitted into the conference call. The second portion of text is not converted into a voice signal, as the second user is determined not to be relevant.

SPEAKER PRIORITY FOR CONFERENCE CALLS

An example method for managing a conference call includes detecting multiple concurrent audio signals. The multiple concurrent audio signals include a first audio signal from a first participant on a conference call and a second audio signal from a second participant on the conference call. The example method also includes comparing a first cumulative time period in which the first participant has spoken to a second cumulative time period in which the second participant has spoken on the conference call. The example method further includes selecting, based on the comparison, one of the first participant and the second participant as a lower ranked participant. The example method also includes providing an indication to the lower ranked participant to yield speaking in favor of another participant on the conference call. 1. A method for managing a conference call , comprising:detecting, by an input/output (I/O) module, multiple concurrent audio signals, the multiple concurrent audio signals including a first audio signal from a first communications device coupled to a conference call and a second audio signal from a second communications device coupled to the conference call, a first participant being located remotely from the second participant and using the first communications device to communicate with one or more participants on the conference call, and a second participant using the second communications device to communicate with one or more participants on the conference call;tracking, by a time keeping module, a first cumulative time period in which audio signals from the first communications device are received and a second cumulative time period in which audio signals from the second communications device are received, the time keeping module being remote from at least one of the first and second communications devices;determining whether the multiple concurrent audio signals are detected for a threshold period of time; and comparing the first cumulative ...

SYSTEM AND METHOD FOR MULTI-PARTY COMMUNICATION OF TRENDING TOPICS

A system and method for multi-party communication includes determining a trending common identifier within a media platform and generating a communication channel within a communication platform based on the common identifier. The method and system includes, in response to a join request from each of a plurality of users, connecting the plurality of users to the communication channel, where the communication channel is an open communication channel for active audio communication between the plurality of users across the communication platform and facilitating audio communication between the plurality of users, including providing a visual display on the communication platform, including visual display of the active audio communication across the communication platform. 1. A method for multi-party communication comprising:determining an identifier identifying a trending topic on a social networking system, wherein the trending topic is a topic of a plurality of social media posts posted on the social networking system;generating a communication channel within a communication platform based on the common identifier, the communication channel identified by the trending topic;in response to a join request from each of a plurality of users, connecting the plurality of users to the communication channel identified by the trending topic;facilitating audio communication between the plurality of users, including providing a visual display on the communication platform, including visual display of the active audio communication across the communication platform; andmonitoring the social network system for determining if the identifier continues to identify the trending topic on the social network system to maintain the communication channel while the identifier identifies the trending topic.2. The method of claim 1 , wherein the media platform is a social media platform for users communicating short messages.3. The method of claim 1 , wherein the common identifier is a ...

SYSTEM AND METHOD FOR MULTI-PARTY COMMUNICATION

A system and method provides for multi-party communication including engaging a network communication engine from a mobile communication interface and providing a user identification and a location indicator to the network communication engine. The method and system further includes receiving in the mobile communication interface a plurality of communication channels based on at least one of: the user identification and the location indicator and in response to a user selection of a selected communication channel, transmitting a channel engagement request to the network communication engine requesting connection of the user to the selected communication channel. 1. A method for multi-party communication comprising:connecting to a network communication engine from a mobile communication interface operating on a mobile device;providing a user identification to the network communication engine from the mobile device;displaying in the mobile communication interface a plurality of communication channels based on the user identification and at least one user preference associated with the user identification, wherein the communication channels are facilitated by the network communication engine for active audio communication between the user identified by the user identification and additional users;connecting the user to a selected communication channel, the selected communication channel having a channel moderator therein;facilitating communication between the user and a plurality of the additional users across the selected communication channel, wherein the channel moderator designates the user communication mode including at least one of: open mode, a touch-to-talk mode, and a listen-only mode;in response to a channel moderator instruction, establishing an original mode for the user entering the communication channel;receiving another channel moderator instruction for changing the communication mode for the user; andupdating the user communication with the channel ...

Method for Improving Perceptual Continuity in a Spatial Teleconferencing System

The present document relates to audio conference systems. In particular, the present document relates to improving the perceptual continuity within an audio conference system. According to an aspect, a method for multiplexing first and second continuous input audio signals is described, to yield a multiplexed output audio signal which is to be rendered to a listener. The first and second input audio signals () are indicative of sounds captured by a first and a second endpoint (), respectively. The method comprises determining a talk activity () in the first and second input audio signals (), respectively; and determining the multiplexed output audio signal based on the first and/or second input audio signals () and subject to one or more multiplexing conditions. The one or more multiplexing conditions comprise: at a time instant, when there is talk activity () in the first input audio signal (), determining the multiplexed output audio signal at least based on the first input audio signal (); at a time instant, when there is talk activity () in the second input audio signal (), determining the multiplexed output audio signal at least based on the second input audio signal (); and at a silence time instant, when there is no talk activity () in the first and in the second input audio signals (), determining the multiplexed output audio signal based on only one of the first and second input audio signals (). 2. The method of claim 1 , wherein determining the multiplexed output audio signal comprisesapplying a first time-dependent to the first input audio signal, to yield a first weighted audio signal; wherein the first gain is greater than zero, at time instants when there is talk activity in the first input audio signal;applying a second time-dependent gain to the second input audio signal, to yield a second weighted audio signal; wherein the second gain is greater than zero, at time instants when there is talk activity in the second input audio signal; anddetermining ...

Dynamically Controlled Aspect Ratios for Communication Session Video Streams

The disclosed techniques improve user engagement and promote efficient use of computing resources by providing dynamically controlled aspect ratios for communication session renderings based on a physical orientation of a device. In some configurations, a system can select a first aspect ratio for individual video streams of a communication session when a device is in a first orientation, e.g., a portrait orientation. In addition, the system can select a second aspect ratio for the individual video streams when the device is in a second orientation, e.g., a landscape orientation. In some configurations, the first aspect ratio can be greater than the second aspect ratio, or the aspect ratios can be selected based on a target aspect ratio, which can be adjusted over time. By dynamically selecting an aspect ratio for individual stream renderings, screen space of a device can be optimized while the device is held in various physical orientations.

Spectral shaping for audio mixing

Techniques are described herein that are capable of spectrally shaping audio signal(s) for audio mixing. Spectrally shaping an audio signal means modifying a frequency spectrum of the audio signal. A frequency spectrum of an audio signal is a representation of the audio signal in the frequency domain. For instance, a frequency spectrum may be represented using multiple frequency bands. The frequency spectrum may be modified by modifying characteristic(s) (e.g., magnitude, phase, etc.) one or more of the frequency bands.

Audio data processing method, server, client and server, and storage medium

A server, system, and client are provided. The server receives audio data from one or more clients; receives an audio data request from a requesting client, and obtains, from the audio data request, a mixing manner of audio data requested by the requesting client; preprocesses audio data of a client of the one or more clients that corresponds to the audio data request using the obtained mixing manner; and sends the preprocessed audio data to the requesting client.

METHOD AND SYSTEM FOR CONTROLLING COMMUNICATIONS FOR VIDEO/AUDIO-CONFERENCING

The present invention relates to a method for controlling communications for an online meeting. The method includes, at each of a plurality of user devices, receiving audio from one of a plurality of participants via a microphone, at least one processor coordinating transmission of the audio from each of the plurality of user devices to the other user devices via a communications system, the processors progressing through multiple stages for the online meeting where each stage is associated with audio configuration data, and at each stage, the at least one processor automatically muting audio received from one or more of the plurality of user devices in accordance with the audio configuration data associated with that stage such that the muted audio is not generated at the other user devices. Other systems and methods for controlling communications and managing online meetings are disclosed. 1. A method of controlling communications for an online meeting , including:a. at least one processor coordinating transmission of audio from each of the plurality of user devices to the other user devices via a communications system;b. the at least one processor progressing through multiple stages for the online meeting; wherein each stage is associated with audio configuration data; andc. at each stage, the at least one processor automatically muting audio received from one or more of the plurality of user devices in accordance with the audio configuration data associated with that stage such that the muted audio is not generated at the other user devices;wherein the audio is received at each of the plurality of user devices from one of a plurality of participants via a microphone.2. A method as claimed in claim 1 , wherein the audio configuration data defines when and how muting is to be applied during the stage.3. A method as claimed in claim 1 , wherein a user interface is displayed on at least some of the user devices claim 1 , the user interface comprising an audio ...

Prioritization of speakers in a hearing device system

Systems and methods for controlling hearing devices are disclosed. Each hearing device is worn by a user, the hearing devices being communicatively interconnected with each other for data exchange to form a hearing device system. A method includes acquiring a sound signal of each user with a microphone of the hearing device of the user; determining from the sound signal a speaker signal, which indicates, whether the user is a speaking user; determining a priority score for each user from the speaker signal of the user; outputting sound signals from speaking users to the users via the hearing devices of the users, such that sound signals from speaking users with a priority score smaller than a threshold are muted.

Methods of generating visual composition for multimedia conference event

FIELD: radio engineering, communication. SUBSTANCE: invention relates to a multimedia conferencing system which displays different types of media content using various graphical user interface windows or views. An apparatus for managing meetings in a multimedia conferencing system comprises a visual composition component operative to generate a visual composition for a multimedia conference event. The visual composition component may comprise a video decoder module operative to decode multiple media streams for a multimedia conference event; an active participant detector module operative to detect a participant in a decoded media stream as an active speaker; a media stream manager module operative to map the decoded media stream with the active speaker to an active display frame and the other decoded media streams to non-active display frames; and a visual composition generator module operative to generate a visual composition with a participant list having the active and non-active display frames positioned in a predetermined order. EFFECT: improved identification of different types participants of a meeting in a virtual environment. 14 cl, 6 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 518 402 C2 (51) МПК H04N 7/15 (2006.01) H04L 12/18 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ (21)(22) Заявка: ИЗОБРЕТЕНИЯ К ПАТЕНТУ 2010133959/07, 29.01.2009 (24) Дата начала отсчета срока действия патента: 29.01.2009 Приоритет(ы): (30) Конвенционный приоритет: (43) Дата публикации заявки: 20.02.2012 Бюл. № 5 (73) Патентообладатель(и): МАЙКРОСОФТ КОРПОРЕЙШН (US) R U 14.02.2008 US 12/030,872 (72) Автор(ы): ТХАККАР Пулин (US), СИНГХ Ноор-Е-Гаган (US), ДЖАИН Стути (US), ИКС (US), БХАТТАЧАРДЖИ Авронил (US) (45) Опубликовано: 10.06.2014 Бюл. № 16 (85) Дата начала рассмотрения заявки PCT на национальной фазе: 13.08.2010 C 2 C 2 2004230651 A1, 2004-11-18 . US 2004008249 A1, 2004-01-15 . US 2005078171 A1, 2005-04-14 . US 2007294346 A1, 2007-12-20 . US ...

Centrally controlling communication at a venue

One example may include a method that includes initiating an audio recording to capture audio data, comparing the audio data received from a microphone of a mobile device to an audio data range, determining whether the audio data is above an optimal level based on a result of the comparison, and queuing the audio data in an audio data queue when the audio data is above the optimal level.

Method and apparatus for suppressing music on hold

The subject invention is directed to a method and system for removing music or other unwanted noise from a conference call particularly when the music or other unwanted noise is not directly under the control of the telecommunication equipment the user is attached to and using. The method and system detects the unwanted sound, deactivates the sound source or disconnects from the line carrying the sound, monitors the disconnected line for voice energy, and reconnects the line when voice energy is detected. This capability provides the telecommunications equipment user with control over unwanted music sources being provided by equipment located either in the private or public networks.

Centrally controlling communication at a venue

One example may include a method that includes receiving, at a server, a data set from one or more mobile devices located in a presentation space, combining the received data set with additional data to create a combined data set, creating a presentation signal based on the combined data set, subtracting a portion of one or more of the data set and the additional data set from the combined data set to create a modified presentation signal, forwarding the modified presentation signal to one or more of a display and a loudspeaker located in the presentation space, and playing the modified presentation signal via one or more of the loudspeaker and the display device.

Teleforum apparatus and method

A method and apparatus of providing a teleforum. The method includes generating simultaneous outbound invitation calls to multiple invitees and transmitting a message to each invitee, inviting him or her to become a teleforum participant. Teleforum participants who accept the invitation are connected to the teleforum in a listen-only mode. During the teleforum, a teleforum participant may request active participation in the teleforum. The method also includes allowing the participant to actively participate in the teleforum. Active participation may include but is not limited to such activities as: speaking to the teleforum, asking a question, responding to a poll, making a donation, providing information, leaving a message, being transferred to a subconference, receiving feedback from participants, leaving a voicemail, volunteering for future activities or receiving media transmitted outside of the teleforum.

Managing users within a group that share a single teleconferencing device

A method for identifying and managing users within a group during a teleconference. The method includes a computer processor determining that a group of users are sharing a first client device to communicate, via a teleconference with one or more other users. The method further includes identifying a first user, of the group of user that are sharing the first client device, that is speaking during a first portion of the teleconference. The method further includes determining an action corresponding to the first user, where an action affects managing content of the teleconference at a client device. The method further includes initiating the determined action corresponding to the first user.

Teleforum apparatus and method

A method and apparatus of providing a teleforum. The method includes generating simultaneous outbound invitation calls to multiple invitees and transmitting a message to each invitee, inviting him or her to become a teleforum participant. Teleforum participants who accept the invitation are connected to the teleforum in a listen-only mode. During the teleforum, a teleforum participant may request active participation in the teleforum. The method also includes allowing the participant to actively participate in the teleforum. Active participation may include but is not limited to such activities as: speaking to the teleforum, asking a question, responding to a poll, making a donation, providing information, leaving a message, being transferred to a subconference, receiving feedback from participants, leaving a voicemail, volunteering for future activities or receiving media transmitted outside of the teleforum.

Webcasting with just-in-time resource provisioning, automated signal acquisition and streaming, and fully-automated event archival

The present invention implements a multimedia business communications platform that enables conference casting over a network. End users or so-called "audience members" participate in conference casts from anywhere on the Internet or behind an enterprise firewall using a standard Web browser running a streaming media player. The conference platform typically includes a number of subsystems: a Web reservation subsystem, a voice server subsystem, a content storage subsystem, a monitoring data subsystem, an attendee access subsystem, and an archive subsystem. Web reservation subsystem provides a mechanism that enables an entity to make an event reservation. Web reservation subsystem typically includes a database and an associated database management system. Voice server subsystem validates a user and converts an incoming telephone signal to an output data stream. Attendee access subsystem responds to attendee requests and returns information that directs an attendee's streaming media player on how to join the conference cast. Content storage subsystem is used to archive the event files. The monitoring data subsystem comprises a set of monitoring agents distributed throughout the network. Using these components, the system implements just-in-time resource provisioning, automated signal acquisition and streaming and automated archiving upon event completion.

Conference bridge system

A conference bridge arrangement is disclosed which removes the necessity for "real time" additions of conferenced data sample sums. A memory store (9) is used containing pre-determined valuations of these data sample sums which may be addressed using the usually companded code of data samples and output as such without code conversions. <IMAGE>

Distinguishing conference participants

Technologies are described herein for distinguishing between conference participants associated with different groups. A communication is received from a participant during a conference. The participant's group affiliation is determined and a representation attribute corresponding to the group is used to render the communication within the conference. As a result, the communications from participants of like groups are displayed with like representation attributes to visually distinguish the communications according to group affiliations. An administrator may designate the representation attributes so that all of the communications from participants of the same group are similarly distinguished without action from the participants. Participants may filter the communications as desired according to the groups and corresponding representation attributes.

Audio conferencing system streaming summed conference signal onto the internet

An audio conferencing system comprises an audio conference mixer (90) that receives digitized audio signals and sums a plurality of the digitized audio signals containing speech to provide a summed conference signal. A transcoder (as) receives and transcodes the summed conference signal to provide a transcoded summed signal that is streamed onto the Internet.

Centrally controlling communication at a venue

One example may include a method that includes receiving, at a presentation server, an audio data signal from a mobile device located in a presentation space, identifying a mobile device identification characteristic of the mobile device based on the received audio data signal, determining a mobile device location via a location determination procedure, and playing the audio signal via a loudspeaker.

A communication device for providing an efficient dormant mode for a group communication network

A method and apparatus for providing an efficient dormant mode for push-to-talk communication devices (102, 104, 106) in a group communication network (100) provides for determining whether the communication device has been inactive for a predetermined first time period and, if so, causing the communication device to enter a control-hold mode, wherein the communication device maintains its dedicated traffic channel. The method further includes the steps of determining whether the communication device has been in the control-hold mode for a predetermined second time period and, if so, causing the communication device to enter a dormant mode, wherein the communication device releases its dedicated traffic channel. The method and apparatus further provides for causing the communication device to cache its service configuration state before entering the dormant mode.

Communication control system

The present invention provides a communication control system which can prevent unnecessary communication connections while ensuring user-friendliness when 1 to N communication is performed. The communication control system judges whether an area code of a mobile terminal and area codes of the other mobile terminals of the group match, and establishes 1 to N communication between the mobile terminal and the mobile terminals of which area codes match. As a result, when the users are contacted to meet, for example, the communication control system can prevent unnecessary communication connections without indiscriminately connecting the mobile terminals of users at distant locations for whom it is difficult to travel for a meeting. In addition, the communication destination mobile terminals are automatically selected based on the area code, user-friendliness can therefore be sufficiently ensured.

Notification of audio failure in a teleconference connection

Verfahren, umfassend: Empfang, an einer Telekonferenzbrücke, einer Mehrzahl von Empfangs-Audio-Signalen, ausgehend von an einer Konferenzverbindung beteiligten Endpunkten; Erzeugung, an der Telekonferenzbrücke, für die Endpunkte, einer Mehrzahl von Übertragungs-Audio-Signalen, die auf mindestens einem der Empfangs-Audio-Signale basieren, Erkennen, ob eines der Empfangs-Audio-Signale aus der Mehrzahl der Empfangs-Audio-Signale aus einem der Übertragungs-Audio-Signale ausgeschlossen ist; Erkennen, im Empfangs-Audio-Signal, ob ein Anwender an dem Endpunkt, der mit dem Empfangs-Audio-Signal assoziiert ist, gerade spricht; und wenn das Empfangs-Audio-Signal als ausgeschlossen erkannt wird und der Anwender als gerade sprechend erkannt wird, Übertragung, an einen der Endpunkte, eines hörbaren Hinweises als Teil der Mehrzahl der Übertragungs-Audio-Signale. A method, comprising: receiving, at a teleconferencing bridge, a plurality of receive audio signals, from endpoints involved in a conference call; Generating, at the teleconferencing bridge, for the endpoints, a plurality of transmit audio signals based on at least one of the receive audio signals, detecting whether one of the receive audio signals from the plurality of receive audio signals is excluded from any of the transmission audio signals; Detecting, in the receive audio signal, whether a user is currently speaking at the endpoint associated with the receive audio signal; and if the receive audio signal is detected as excluded and the user is recognized as being speaking, transmitting to one of the endpoints an audible indication as part of the plurality of transmit audio signals.

Network mute feature in wireless telecommunications systems

A network mute feature deployed in a wireless telecommunications system allows a mobile user to eliminate ambient environmental and wireless transmission noise. Upon activation of the network mute feature, a voice path interconnecting the mobile unit to a party served by the public switched telephone network (PSTN) is opened. A portion of the open voice path is subsequently interconnected to a noise generator so that the PSTN party hears unobtrusive background noise indicating that the mobile user is still "on the line" and can hear transmissions by the party. The PSTN party, however, cannot hear noises associated with mobile user's environment or wireless transmission while the network mute feature is activated. Advantageously, the network mute feature enhances the ability of mobile users to participate in calls in which the quality of transmission (e.g., a conference call) is important.

Method for improving perceptual continuity in a spatial teleconferencing system

The present document relates to audio conference systems. In particular, the present document relates to improving the perceptual continuity within an audio conference system. According to an aspect, a method for multiplexing first and second continuous input audio signals is described, to yield a multiplexed output audio signal which is to be rendered to a listener. The first and second input audio signals ( 123 ) are indicative of sounds captured by a first and a second endpoint ( 120, 170 ), respectively. The method comprises determining a talk activity ( 201, 202 ) in the first and second input audio signals ( 123 ), respectively; and determining the multiplexed output audio signal based on the first and/or second input audio signals ( 123 ) and subject to one or more multiplexing conditions. The one or more multiplexing conditions comprise: at a time instant, when there is talk activity ( 201 ) in the first input audio signal ( 123 ), determining the multiplexed output audio signal at least based on the first input audio signal ( 123 ); at a time instant, when there is talk activity ( 202 ) in the second input audio signal ( 123 ), determining the multiplexed output audio signal at least based on the second input audio signal ( 123 ); and at a silence time instant, when there is no talk activity ( 201, 202 ) in the first and in the second input audio signals ( 123 ), determining the multiplexed output audio signal based on only one of the first and second input audio signals ( 123 ).

Method for managing a conference call in a telephone network

It is disclosed a method of managing, in a telephone network, a conference call among a first terminal and a plurality of terminals, comprising: setting up at least one voice channel for activating said conference call among said first terminal and said plurality of terminals; and causing deactivation of a microphone of at least one terminal of said plurality of terminals.

Electronic accessory incorporating dynamic user-controlled audio muting capabilities, related methods and communications terminal

Electronic accessory device capable of dynamically muting the microphone and/or speaker responsive to predefined user control input, and a communications terminal at least functionally connected to the accessory for the transfer of audio signals and related signaling. Related methods for execution by the accessory and the terminal are presented.

System and method for multi-party communication of trending topics

A system and method for multi-party communication includes determining a trending common identifier within a media platform and generating a communication channel within a communication platform based on the common identifier. The method and system includes, in response to a join request from each of a plurality of users, connecting the plurality of users to the communication channel, where the communication channel is an open communication channel for active audio communication between the plurality of users across the communication platform and facilitating audio communication between the plurality of users, including providing a visual display on the communication platform, including visual display of the active audio communication across the communication platform.

Webcasting with just-in-time resource provisioning, automated signal acquisition and streaming, and fully automated event archiving

The present invention implements a multimedia business communications platform that enables conference casting over a network. End users or so-called "audience members" participate in conference casts from anywhere on the Internet or behind an enterprise firewall using a standard Web browser running a streaming media player. The conference platform typically includes a number of subsystems: a Web reservation subsystem, a voice server subsystem, a content storage subsystem, a monitoring data subsystem, an attendee access subsystem, and an archive subsystem. Web reservation subsystem provides a mechanism that enables an entity to make an event reservation. Web reservation subsystem typically includes a database and an associated database management system. Voice server subsystem validates a user and converts an incoming telephone signal (i.e., the conference call) to an output data stream. Attendee access subsystem responds to attendee requests and returns information (e.g., a data structure such as a stream metafile) that directs an attendee's streaming media player on how to join the conference cast. Content storage subsystem is used to manage event information and, under the control of the archive subsystem, to archive the event files. The monitoring data subsystem comprises a set of monitoring agents distributed throughout the network. Using these components, the system implements just-in-time resource provisioning, automated signal acquisition and streaming, and automated archiving upon event completion.

Call audio mixing processing method and device, storage medium and computer equipment

本申请涉及一种通话音频混音处理方法、装置、计算机可读存储介质和计算机设备，所述方法包括：获取参与通话的通话成员终端发送的通话音频；对各所述通话音频分别进行语音分析，确定与各所述通话成员终端对应的语音活跃度；所述语音活跃度用于反映所述通话成员参与通话的活跃程度；根据所述语音活跃度确定与各所述通话成员终端分别对应的语音调节参数；按照各所述通话成员终端分别对应的语音调节参数，对相应的通话音频进行调节得到调节音频，并基于各所述调节音频进行混音处理得到混合音频。本申请提供的方案可以提高语音通话质量。

Audio conferencing system streaming summed conference signal onto the internet

An audio conferencing system comprises an audio conference mixer (90) that receives digitized audio signals and sums a plurality of the digitized audio signals containing speech to provide a summed conference signal. A transcoder (as) receives and transcodes the summed conference signal to provide a transcoded summed signal that is streamed onto the Internet.

Method, equipment and system for processing multipath audio signal

本发明实施例提供一种多路音频信号处理方法，包括：接收来自与会的会议终端的L路音频信号，并分别进行解码，根据解码得到的数据从与会的会议终端中确定N个待混音的会议终端；选择与会的会议终端中除N个待混音的会议终端外的L－N个编码器中的任一编码器作为共用编码器，对解码后的N个待混音的会议终端的音频信号编码，将编码之后的数据发送至与所述L－N个编码器对应的会议终端；L－N个编码器中除共用编码器外的其余L－N－1个编码器更新其编码状态信息以与上述共用编码器的编码状态信息保持同步。同时公开相应的设备及系统，本发明的实施例可以节约处理器的资源，在处理器资源一定的情况下增加可接入信源的数量。

Ambient sound rendering for online meetings

Techniques of conducting an online meeting involve outputting ambient sound to a participant of an online meeting. Along these lines, in an online meeting during which a participant wears headphones, the participant's computer receives microphone input that contains both speech from the participant and ambient sound that the participant may wish to hear. In response to receiving the microphone input, the participant's computer separates low-volume sounds from high-volume sounds. However, instead of suppressing this low-volume sound from the microphone input, the participant's computer renders this low-volume sound. In most cases, this low-volume sound represents ambient sound generated in the vicinity of the meeting participant. The participant's computer then mixes the low-volume sound with speech received from other conference participants to form output in such a way that the participant may distinguish this sound from the received speech. The participant's computer then provides the output to the participant's headphones.

Apparatus and method for transmission and receipt of conference call roster information via the internet

An apparatus and method is disclosed that allows a participant in a conference call to receive conference call roster information at a telephone display unit, at a computer display or at both types of display. The conference call information is updated during the call and includes identification of the speaker during a conference call. The conference callers are connected together and identified either by standard caller identification or by other identification provided by the operator. The conference callers' identifications are then transmitted in two ways. First, a coded signal is sent over the telephone connection to a telephone display unit. Second, an information packet is transmitted over the Internet to a conference call participant's computer. A server computer monitors the conference call and the identification of parties involved is updated and parties are added or disconnected. The speaking party is identified in one of two ways. First, during the call, the speaker's telephone line will have a higher transmission amplitude than the other telephone lines and the line with the highest transmission amplitude will be identified by the server computer. Second, voice identification data is gathered before the conference call and stored in the server computer for use in identifying the speaker.

Active speaker identification

Изобретение относится к компьютерной технике и может использоваться для идентификации клиентов во время аудио-события. Технический результат состоит в повышении точности идентификации аудио-клиентов. Для этого медиа-сервер может упорядочивать предоставляющих аудио клиентов, основываясь на входном уровне. Идентификатор может быть ассоциирован с клиентом для идентификации этого клиента, предоставляющего ввод в пределах события. Упорядоченные клиенты могут быть включены в список, который может быть вставлен в заголовок пакета, переносящего аудио-контент. 3 н. и 17 з.п. ф-лы, 4 ил.

Active speaker identification

오디오 이벤트에서 클라이언트들을 식별하기 위한 프로시저들(procedures)이 기술된다. 예에서, 미디어 서버는 입력 레벨에 기초하여 오디오를 제공하는 클라이언트들을 순서화할 수 있다. 이벤트 내의 입력을 제공하는 클라이언트를 식별하기 위해 식별자가 클라이언트와 연관될 수 있다. 순서화된 클라이언트들은 오디오 컨텐츠를 반송하는 패킷 헤더에 삽입될 수 있는 리스트에 포함될 수 있다. Procedures for identifying clients in an audio event are described. In the example, the media server may order clients that provide audio based on the input level. An identifier may be associated with the client to identify the client providing input in the event. Ordered clients can be included in a list that can be inserted into the packet header carrying the audio content. 미디어 회의, 오디오, 클라이언트, 프로토콜 Media Conference, Audio, Client, Protocol

Method and apparatus for suppressing music on hold

The subject invention is directed to a method and system for removing music or other unwanted noise from a conference call particularly when the music or other unwanted noise is not directly under the control of the telecommunication equipment the user is attached to and using. The method and system detects the unwanted sound, deactivates the sound source or disconnects from the line carrying the sound, monitors the disconnected line for voice energy, and reconnects the line when voice energy is detected. This capability provides the telecommunications equipment user with control over unwanted music sources being provided by equipment located either in the private or public networks.

Be used to the multimedia conferencing incident to generate the synthetic technology of vision

描述了用于为多媒体会议事件生成视觉合成的技术。一种装置可包括可用于为多媒体会议事件生成视觉合成的视觉合成组件。该视觉合成组件可包括视频解码器模块，该视频解码器模块可用于解码多媒体会议事件的多个媒体流；活跃说话者检测器模块，该活跃说话者检测器模块可用于将已解码媒体流中的参与者检测为活跃说话者；媒体流管理器模块，该媒体流管理器模块可用于将具有活跃说话者的已解码媒体流映射到活动显示帧并将其他已解码媒体流映射到非活动显示帧；以及视觉合成生成器模块，该视觉合成生成器模块可用于生成具有以预定次序定位的活动显示帧和非活动显示帧的参与者名单的视觉合成。对其它实施例也予以描述并要求保护。

Controller for providing an efficient dormant mode for a group communication network

A method and apparatus for providing an efficient dormant mode for push-to-talk communication devices in a group communication network provides for determining whether the net has been inactive for a predetermined first time period and, if so, causing each member of the net to enter a control-hold mode, wherein each member of the net maintains its dedicated traffic channel. The method further includes the steps of determining whether each member of the net has been in the control-hold mode for a predetermined second time period and, if so, causing each member of the net to enter a dormant mode, wherein each member of the net releases its dedicated traffic channel. The method and apparatus further provides for causing each member of the net to cache its service configuration state before entering the dormant mode.

System and process for mass telephony conference call

An automated teleconferencing control system and method reduces conference call setup time by simultaneously out dialing to multiple conferees through a plurality of parallel telecommunications resources. The teleconferencing control system includes the feature of automatically gathering statistics of each connection attempt, as well as many other call aspects, using these statistics to automatically improve the efficiency of subsequent conference call initiations. Gathered statistics are also used to automatically configure the system to cause all of the conferees to be connected to a conference call at substantially the same time. The selection of which telecommunications resources are used for contacting each target conferee is made based on criteria that include the telephone number or Internet address of the target conferee's telephone.

Teleconference volume level monitoring and feedback on the volume level

Prioritization of speakers in a hearing device system

Systems and methods for controlling hearing devices are disclosed. Each hearing device is worn by a user, the hearing devices being communicatively interconnected with each other for data exchange to form a hearing device system. A method includes acquiring a sound signal of each user with a microphone of the hearing device of the user; determining from the sound signal a speaker signal, which indicates, whether the user is a speaking user; determining a priority score for each user from the speaker signal of the user; outputting sound signals from speaking users to the users via the hearing devices of the users, such that sound signals from speaking users with a priority score smaller than a threshold are muted.

Systems and methods for improving audio conferencing services

Systems and methods are disclosed herein for improving audio conferencing services. One aspect relates to processing audio content of a conference. A first audio signal is received from a first conference participant, and a start and an end of a first utterance by the first conference participant are detected from the first audio signal. A second audio signal is received from a second conference participant, and a start and an end of a second utterance by the second conference participant is detected from the second audio signal. The second conference participant is provided with at least a portion of the first utterance, wherein at least one of start time, start point, and duration is determined based at least in part on the start, end, or both, of the second utterance.

Audio data processing method, server, client and server, and storage medium

A server, system, and client are provided. The server receives audio data from one or more clients; receives an audio data request from a requesting client, and obtains, from the audio data request, a mixing manner of audio data requested by the requesting client; preprocesses audio data of a client of the one or more clients that corresponds to the audio data request using the obtained mixing manner; and sends the preprocessed audio data to the requesting client.

Method and system for transmitting caller identification information in a conference call

A method and system for transmitting caller identification information during a conference call is disclosed. When a caller enters into a phone conference, the endpoint device he is using can subscribe to a notification service provided by the conference server. Thereafter, the conference server sends out a message with the phone number (or other indication) of the active endpoint device. Each time another endpoint device in conference call becomes the active device (e.g., someone using that endpoint device begins speaking), the conference server senses this and sends a new message to the endpoint devices indicating this change.

Active speaker identification

Procedures for identifying clients in an audio event are described. In an example, a media server may order clients providing audio based on the input level. An identifier may be associated with the client for identifying the client providing input within the event. The ordered clients may be included in a list which may be inserted into a packet header carrying the audio content.

Conference screening

A method and apparatus of providing a teleforum. The method includes generating simultaneous outbound invitation calls to multiple invitees and transmitting a message to each invitee, inviting him or her to become a teleforum participant. Teleforum participants who accept the invitation are connected to the teleforum in a listen-only mode. During the teleforum, a teleforum participant may request active participation in the teleforum. The method also includes allowing the participant to actively participate in the teleforum. Active participation may include but is not limited to such activities as: speaking to the teleforum, asking a question, responding to a poll, making a donation, providing information, leaving a message, being transferred to a subconference, receiving feedback from participants, leaving a voicemail, volunteering for future activities or receiving media transmitted outside of the teleforum.

Method, device, and system for processing multi-channel audio signals

A method for processing multi-channel audio signals includes receiving L channel audio signals from participating conference terminals, decoding the L channel audio signals, and determining N conference terminals whose audio signals are to be mixed from the participating conference terminals according to the data obtained through decoding; selecting an encoder as a common encoder from L−N encoders of the participating conference terminals except the N conference terminals whose audio signals are to be mixed, encoding decoded audio signals of the N conference terminals whose audio signals are to be mixed, and sending encoded data to conference terminals corresponding to the L−N encoders (indicating that the number of encoders is equal to the value of L minus N); and updating encoding status information of L−N−1 encoders except the common encoder to synchronize with encoding status information of the common encoder.

Techniques to generate a visual composition for a multimedia conference event

Techniques to generate a visual composition for a multimedia conference event are described. An apparatus may comprise a visual composition component operative to generate a visual composition for a multimedia conference event. The visual composition component may comprise a video decoder module operative to decode multiple media streams for a multimedia conference event, an active speaker detector module operative to detect a participant in a decoded media stream as an active speaker, a media stream manager module operative to map the decoded media stream with the active speaker to an active display frame and the other decoded media streams to non-active display frames, and a visual composition generator module operative to generate a visual composition with a participant roster having the active and non-active display frames positioned in a predetermined order. Other embodiments are described and claimed.

Method of, and system for, webcasting with just-in-time resource provisioning, automated telephone signal acquistion and streaming, and fully-automated event archival

The present invention implements a multimedia business communications platform that enables conference casting over a network. End users or so-called “audience members” participate in conference casts from anywhere on the Internet or behind an enterprise firewall using a standard Web browser running a streaming media player. The conference platform typically includes a number of subsystems: a Web reservation subsystem, a voice server subsystem, a content storage subsystem, a monitoring data subsystem, an attendee access subsystem, and an archive subsystem. Web reservation subsystem provides a mechanism that enables an entity to make an event reservation. Web reservation subsystem typically includes a database and an associated database management system. Voice server subsystem validates a user and converts an incoming telephone signal (i.e., the conference call) to an output data stream. Attendee access subsystem responds to attendee requests and returns information (e.g., a data structure such as a stream metafile) that directs an attendee's streaming media player on how to join the conference cast. Content storage subsystem is used to manage event information and, under the control of the archive subsystem, to archive the event files. The monitoring data subsystem comprises a set of monitoring agents distributed throughout the network. Using these components, the system implements just-in-time resource provisioning, automated signal acquisition and streaming, and automated archiving upon event completion.

Conference call peer-driven degraded line detection and removal

An embodiment of this invention is about a method of determining which line is causing noise based on peer feedback. Instead of relying on special monitors or algorithms to determine which line is causing the noise, the participants of the call are paired up and asked to identify themselves and to observe the other parties audio for line noise. One system capitalizes on the fact the ability of the caller to determine if the line is producing noise is in most cases far superior to that of the conference system. One method does not require evaluation to be done by the “controlling leg” (i.e. the moderator). That is, there is no need for a centralized authority which makes the decisions on line noise. Instead, each participant is asked to evaluate another participant.

Telephony/conference activity presence state

A presence system and method are described herein which infer a person's level of participation in a telephone call (e.g., multi-party conference call, one-on-one telephone call, collaboration session). In one embodiment, the presence system collects presence information about one or more activities of a person while they are also participating in a telephone call. The presence system aggregates and analyzes the collected presence information to determine a telephony/conference activity presence state which is then published to watchers and/or an automated interaction management system in order to inform them if the person happens to be an “active participant” or a “passive participant” in the telephone call. The watchers and/or the automated interaction management system can then use this information as a guide to determine if the person is likely to be able to respond to another interaction while that person is also participating in the telephone call. The inference is that a person who is “passive participant” is substantially more likely to engage in an IM or even a second phone interaction with another person (e.g., watcher) then if that person was an “active participant”.

Conference system and process for voice activation in the conference system

Conference systems are used for example in discussions and usually comprise a plurality of delegate units with microphones, whereby in a discussion each discussion participant uses his own delegate unit. Usually the delegate units have a switch or the like, that allows the participant in front of the delegate unit to request, that his microphone is activated, so that the speech of the participant is input in the conference system and amplified by the conference system. A conference system comprising ( 1 ) a plurality of delegate units ( 2 ), each delegate unit ( 2 ) having a microphone ( 5 ) for receiving an audio signal from a surrounding, a central service module ( 3 ) handling a plurality of contribution channels, whereby the audio output of the contribution channels contribute to an amplified audio output of the conference system ( 1 ) is proposed, whereby each delegate unit ( 2 ) is adapted to send a request for a contribution channel commit to the central service module ( 3 ), whereby the central service module ( 2 ) is adapted to grant the request and to allocate a contribution channel to the requesting delegate unit ( 2 ), thus setting the requesting delegate unit i in an active state A, whereby the delegate unit ( 2 ) is adapted to trigger the request by voice activation, whereby the request is triggered in case at least a first trigger condition is fulfilled defining that the audio signal level of one of the delegate units ( 2 ) as a possible requesting delegate unit i is higher than an individual test value for each other delegate unit ( 2 ) in the active state A, whereby the individual test value is an estimated audio signal level of the possible requesting unit i resulting from an audio or speech signal provided to the other active delegate units ( 2 ).

Call audio mixing processing

A call audio mixing processing method is provided. In the method, call audio streams from terminals of call members participating in a call are obtained. Voice analysis is performed on the call audio streams to determine voice activity corresponding to each of the terminals. The voice activity of the terminals indicate activity levels of the call members participating in the call. According to the voice activity of the terminals, respective voice adjustment parameters corresponding to the terminals are determined. According to the respective voice adjustment parameters corresponding to the terminals, the call audio streams of the terminals are adjusted. Further, mixing processing is performed on the adjusted call audio streams to obtain a mixed audio stream.

用于会议室中多个设备的共享扬声器电话系统

提供了用于会议室中多个设备的共享扬声器电话系统。扬声器电话系统由正在使用web会议服务的物理集会中的参与者的多个参与者设备共享。从参与者中标识活跃发言者。活跃发言者的参与者设备被切换，使得扬声器电话系统接收并呈现活跃发言者的音频。启用活跃发言者的参与者设备的视频，使得web会议服务向参与者设备显示该视频。

Dynamically controlled aspect ratios for communication session video streams

The disclosed techniques improve user engagement and promote efficient use of computing resources by providing dynamically controlled aspect ratios for communication session renderings based on a physical orientation of a device. In some configurations, a system can select a first aspect ratio for individual video streams of a communication session when a device is in a first orientation, e.g., a portrait orientation. In addition, the system can select a second aspect ratio for the individual video streams when the device is in a second orientation, e.g., a landscape orientation. In some configurations, the first aspect ratio can be greater than the second aspect ratio, or the aspect ratios can be selected based on a target aspect ratio, which can be adjusted over time. By dynamically selecting an aspect ratio for individual stream renderings, screen space of a device can be optimized while the device is held in various physical orientations.

Dynamically controlled aspect ratios for communication session video streams

The disclosed techniques improve user engagement and promote efficient use of computing resources by providing dynamically controlled aspect ratios for communication session renderings based on a physical orientation of a device. In some configurations, a system can select a first aspect ratio for individual video streams of a communication session when a device is in a first orientation, e.g., a portrait orientation. In addition, the system can select a second aspect ratio for the individual video streams when the device is in a second orientation, e.g., a landscape orientation. In some configurations, the first aspect ratio can be greater than the second aspect ratio, or the aspect ratios can be selected based on a target aspect ratio, which can be adjusted over time. By dynamically selecting an aspect ratio for individual stream renderings, screen space of a device can be optimized while the device is held in various physical orientations.

Dynamically controlled aspect ratios for communication session video streams

The disclosed techniques improve user engagement and promote efficient use of computing resources by providing dynamically controlled aspect ratios for communication session renderings based on a physical orientation of a device. In some configurations, a system can select a first aspect ratio for individual video streams of a communication session when a device is in a first orientation, e.g., a portrait orientation. In addition, the system can select a second aspect ratio for the individual video streams when the device is in a second orientation, e.g., a landscape orientation. In some configurations, the first aspect ratio can be greater than the second aspect ratio, or the aspect ratios can be selected based on a target aspect ratio, which can be adjusted over time. By dynamically selecting an aspect ratio for individual stream renderings, screen space of a device can be optimized while the device is held in various physical orientations.

Systems and methods for filtering unwanted sounds from a conference call

To filter unwanted sounds from a conference call, a voice profile of a first user is generated based on a first voice signal captured by a media device during a first conference call. The voice profile may be generated by identifying a base frequency of the first voice signal and determining a plurality of voice characteristics, such as pitch, intonation, accent, loudness, and speech rate. These data may be stored in association with the first user. During a second conference call, a second voice signal captured by the media device is analyzed to determine, based on the voice profile of the first user, whether the second voice signal includes the voice of a second user. If so, the second voice signal is prevented from being transmitted into the conference call. A voice profile of the second user may be generated from the second voice signal for future use.

Systems and methods for filtering unwanted sounds from a conference call

To filter unwanted sounds from a conference call, a voice profile of a first user is generated based on a first voice signal captured by a media device during a first conference call. The voice profile may be generated by identifying a base frequency of the first voice signal and determining a plurality of voice characteristics, such as pitch, intonation, accent, loudness, and speech rate. These data may be stored in association with the first user. During a second conference call, a second voice signal captured by the media device is analyzed to determine, based on the voice profile of the first user, whether the second voice signal includes the voice of a second user. If so, the second voice signal is prevented from being transmitted into the conference call. A voice profile of the second user may be generated from the second voice signal for future use.

Systems and methods for filtering unwanted sounds from a conference call using voice synthesis

To filter unwanted sounds from a conference call, a first voice signal is captured by a first device during a conference call and converted into corresponding text, which is then analyzed to determine that a first portion of the text was spoken by a first user and a second portion of the text was spoken by a second user. If the first user is relevant to the conference call while the second user is not, the first voice signal is prevented from being transmitted into the conference call, the first portion of text is converted into a second voice signal using a voice profile of the first user to synthesize the voice of the first user, and the second voice signal is then transmitted into the conference call. The second portion of text is not converted into a voice signal, as the second user is determined not to be relevant.

Speaker Identification for Use in Multi-Media Conference Call System

Methods for operating a meeting coordinator to detect which participant is speaking during a teleconference meeting using a teleconferencing system, having corresponding mobile electronic devices and computer-readable media, comprise: accessing calendaring information concerning the teleconference meeting, the calendaring information including a time of the teleconference meeting, a meeting location, identities of meeting invitees, and contact information for the meeting invitees; automatically setting up the teleconference meeting using the calendaring information to enable the teleconferencing system to connect remotely-located meeting invitees; generating a roster of meeting participants comprising at least some of the meeting invitees; tracking a participation status for the meeting participants; and accessing prerecorded unique digital voice signatures for the meeting invitees.

Leveraging a network of microphones for inferring room location and speaker identity for more accurate transcriptions and semantic context across meetings

Conference call system, method, and computer program product

A conference call system comprises an input interface for receiving during a conference call at least two input streams of audio signal, each from another source. A selection unit is connected to the input interface, for selecting a number of dominant speaker streams out of the input streams, the number being less than or equal to a maximum number of dominant speakers value and each of the dominant speaker streams representing speech from a respective dominant speaker. A mixer is connected to the selection unit, for mixing the selected streams into an output stream. The conference call system comprises an output interface for outputting the output stream and a selection control unit connected to the selection unit and the input interface, for dynamically setting, during the conference call, the maximum number of dominant speakers value based on dynamics of the conference call.

会议电话系统、方法以及计算机程序产品

一种会议电话系统，该会议电话系统包括用于在会议电话期间接收音频信号的至少两个输入流的输入接口，每个输入流都来自另一信源。选择单元被连接到该输入接口，用于从输入流中选择多个主扬声器流，该数目小于或等于主扬声器值的最大数目，并且主扬声器流中的每一个都表示来自相应的主扬声器的话音。混合器被连接到该选择单元，用于将所选择的流混合到输出流中。该会议电话系统包括用于输出该输出流的输出接口和连接到该选择单元和输入接口的选择控制单元，用于在会议电话期间基于会议电话的动态来动态地设置主扬声器值的最大数目。

Speaker identification for use in multi-media conference call system

Methods for operating a meeting coordinator to detect which participant is speaking during a teleconference meeting using a teleconferencing system, having corresponding mobile electronic devices and computer-readable media, comprise: accessing calendaring information concerning the teleconference meeting, the calendaring information including a time of the teleconference meeting, a meeting location, identities of meeting invitees, and contact information for the meeting invitees; automatically setting up the teleconference meeting using the calendaring information to enable the teleconferencing system to connect remotely-located meeting invitees; generating a roster of meeting participants comprising at least some of the meeting invitees; tracking a participation status for the meeting participants; and accessing prerecorded unique digital voice signatures for the meeting invitees.

Hearing assistance system in a multi-talker acoustic network

There is provided a multi-talker acoustic network system for providing hearing assistance to a user, comprising at least two table microphone units (10, 12) for capturing audio signals from a speaker's voice, each comprising a microphone arrangement (30) having an omnidirectional characteristic, a VAD (32) for detecting voice activity of the microphone arrangement and a transmitter (36) for transmitting the captured audio signals via a wireless audio link (40), a control unit (34) for selecting one of tire table microphone units as the presently active microphone unit, wherein the control unit is configured to select, in case that at a time only for one of the table microphone units voice activity is detected, that one of the table microphone units as the presently active microphone unit, and to select, in case that at a time for more than one of the table microphone units voice activity is detected, at least for a certain time period that one of the table microphone units as the presently active microphone unit which has detected the voice activity first as the presently active microphone unit, wherein the system is configured to maintain the selection of the presently active microphone unit with a release time, and a hearing assistance device (20, 22) to be worn by the user, comprising a receiver unit (42) for receiving audio signals captured by the presently active microphone unit and an output transducer (44) for stimulation of the user's hearing according to the received audio signals, wherein the system is configured to prevent audio signals of the table microphone unit(s) not being the presently active microphone unit from being supplied to the output transducer.

간이 교환기의 회의 통화 성립 방법

본 발명은 간이 교환기의 회의 통화 성립에 관한 것으로, 특히 회의 주재자가 모든 회의 참가자들을 지정해 동시에 자동으로 호출하고 음성 서비스 장치를 이용해 회의 시작을 알리며 모든 회의 참가자가 호출 응답해 회의 참가 여부 확인 방법을 회의 주재자에게 제공하도록 한 간이 교환기의 회의 통화 성립 방법에 관한 것이다. 종래 간이 교환기의 회의 통화 성립 방법은 회의 호출 시에 회의 참가자를 한 명씩 호출해서 회의 참가를 알려 줘야 하고 회의 통화가 시작할 때까지 호출된 회의 참가자는 대기해야 할 뿐만 아니라 회의 통화 중에 회의 주재자가 회의 참가자의 참가 상태를 알 수 없는 문제점이 있었다. 본 발명에 의해 간이 교환기의 회의 통화 성립에 있어 회의 참여할 모든 회의 참가자들을 미리 지정한 후에 자동으로 동시에 호출하고 음성 서비스 장치 또는 문자 서비스 장치를 이용해 회의 시작을 알리며 모든 회의 참가자의 회의 호출 응답에 대한 유동 버튼의 LED로서 회의 참가 상태를 확인할 수 있으므로, 통화 성립의 시간 및 과정을 간략화하고 회의 찹가자들에 대한 회의 참가 상태를 쉽게 확인할 수 있다.

Conference calls

Audio data and associated classification data from a plurality of user devices as part of a conference call is provided, the plurality of user devices including at least a first user device and a second user device. The classification data is indicative of one of a plurality of predetermined sound types represented by the associated audio data in a current time frame. Determining that the audio data provided by the first user device is associated with a prioritised dominant speaker. Controlling transmission of the audio data as part of the conference call and preventing transmission of audio data provided by at least the second user device based on its associated classification data being indicative of a restricted sound type for example silence, background noise, coughing, emotive sound, singing or interjections type of sound and the first user device being associated with the dominant speaker.

Conference system and process for voice activation in the conference system

Conference systems are used for example in discussions and usually comprise a plurality of delegate units with microphones, whereby in a discussion each discussion participant uses his own delegate unit. Usually the delegate units have a switch or the like, that allows the participant in front of the delegate unit to request, that his microphone is activated, so that the speech of the participant is input in the conference system and amplified by the conference system. A conference system comprising (1) a plurality of delegate units (2), each delegate unit (2) having a microphone (5) for receiving an audio signal from a surrounding, a central service module (3) handling a plurality of contribution channels, whereby the audio output of the contribution channels contribute to an amplified audio output of the conference system (1) is proposed, whereby each delegate unit (2) is adapted to send a request for a contribution channel commit to the central service module (3), whereby the central service module (2) is adapted to grant the request and to allocate a contribution channel to the requesting delegate unit (2), thus setting the requesting delegate unit i in an active state A, whereby the delegate unit (2) is adapted to trigger the request by voice activation, whereby the request is triggered in case at least a first trigger condition is fulfilled defining that the audio signal level of one of the delegate units (2) as a possible requesting delegate unit i is higher than an individual test value for each other delegate unit (2) in the active state A, whereby the individual test value is an estimated audio signal level of the possible requesting unit i resulting from an audio or speech signal provided to the other active delegate units (2).

会議システム及び会議システムにおけるボイスアクティベーションのための処理方法

会議システムは、例えばディスカッションのために用いられ、一般に、それぞれマイクロフォンを備えた複数のデリゲートユニットを備えており、ディスカッション中、ディスカッション参加者は各々、自分用のデリゲートユニットを使用する。通常、デリゲートユニットはスイッチ或いは同様の部材を備えており、これによってデリゲートユニットの前に位置する参加者は、自分のマイクロフォンの起動を要求することができ、それによって参加者の音声が会議システムへ入力され、会議システムによって増幅される。会議システム（１）は、周囲からオーディオ信号を受信するマイクロフォン（５）を各々有する複数のデリゲートユニット（２）と、複数のコントリビューションチャネルを処理するサービスモジュール（３）を備えている。発言チャネルのオーディオ出力は、会議システム（１）の増幅されたオーディオ出力を構成する。各デリゲートユニット（２）は、コントリビューションチャネルコミット要求を中央サービスモジュール（３）に送信し、中央サービスモジュール（３）はこの要求を許可し、要求送出側のデリゲートユニット（２）に１つのコントリビューションチャネルを割り当て、つまりは要求送出側のユニットをアクティブ状態Ａにセットする。デリゲートユニット（２）は、ボイスアクティベーションによって要求をトリガし、この要求は、少なくとも第１のトリガ条件が満たされた場合にトリガされ、この第１のトリガ条件は、複数のデリゲートユニット（２）のうち、要求送出側の候補デリゲートユニット（ｉ）である１つのデリゲートユニットのオーディオ信号レベルが、アクティブ状態（Ａ）にある他のデリゲートユニット（２）のいずれの個々のテスト値よりも高いことを規定するものであり、個々のテスト値は、他のアクティブなデリゲートユニット（２）へ供給されるオーディオ信号又は音声信号から得られる、要求送出側の候補デリゲートユニット（ｉ）の推定オーディオ信号レベルである。

Systems and methods for filtering unwanted sounds from a conference call using voice synthesis

To filter unwanted sounds from a conference call, a first voice signal is captured by a first device during a conference call and converted into corresponding text, which is then analyzed to determine that a first portion of the text was spoken by a first user and a second portion of the text was spoken by a second user. If the first user is relevant to the conference call while the second user is not, the first voice signal is prevented from being transmitted into the conference call, the first portion of text is converted into a second voice signal using a voice profile of the first user to synthesize the voice of the first user, and the second voice signal is then transmitted into the conference call. The second portion of text is not converted into a voice signal, as the second user is determined not to be relevant.

Steuerung zur bereitstellung eines effizienten ruhemodus für ein gruppenkommunikationsnetzwerk

Steuerung zur bereitstellung eines effizienten schlafmodus für ein gruppenkommunikationsnetzwerk

Effiziente bufferzuteilung für aktuelle und vorhergesagte aktive sprecher in sprechkonferenzsystemen

電話／会議活動プレゼンス状態

【課題】電話通話（例えば、マルチパーティ会議通話、１対１の電話通話、共同セッション）における個人の参加レベルを推測するプレゼンスシステムおよびプレゼンス方法を提供すること。 【解決手段】一実施形態では、プレゼンスシステムは、個人が、電話通話にも参加している間の、個人の１つまたは複数の活動についてのプレゼンス情報を収集する。プレゼンスシステムは、プレゼンス情報を集約し、解析して、電話／会議活動プレゼンス状態を判定し、次に、個人が電話通話において「能動的参加者」であったか、または「受動的参加者」であったかをウォッチャおよび／または自動化された対話管理システムに知らせる。ウォッチャおよび／または自動化された対話管理システムは、次に、その情報を案内として使用して、その個人が電話通話にも参加しながら、別の対話に応答することができる可能性が高いかどうかを判定することができる。 【選択図】図１

电话/会议活动存在状态

这里描述了一种推断电话呼叫(例如，多方会议呼叫、一对一电话呼叫、协同会话)中一个人的参与程度的存在系统和方法。在一个实施方式中，当人在参与电话呼叫的时候存在系统收集和他们的一个或者多个活动相关的存在信息。存在系统综合并分析收集的存在信息以确定电话/会议活动存在状态，然后将其发布给观察者和/或自动交互管理系统以便通知他们这个人在电话呼叫中正处于“主动的参与者”还是“被动的参与者”。然后观察者和/或自动交互管理系统可以利用此信息作为提示来确定这个人在参与电话呼叫的时候是否有可能能够响应另外的交互。推断如下——作为“被动的参与者”的人实际上很有可能参加和其他人(例如，观察者)的IM或者甚至第二电话交互，如果这个人是“主动的参与者”则反之。

Telephony/conference activity presence state

A presence system and method are described herein which infer a person's level of participation in a telephone call (e.g., multi-party conference call, one-on-one telephone call, collaboration session). In one embodiment, the presence system collects presence information about one or more activities of a person while they are also participating in a telephone call. The presence system aggregates and analyzes the collected presence information to determine a telephony/conference activity presence state which is then published to watchers and/or an automated interaction management system in order to inform them if the person happens to be an 'active participant' or a 'passive participant' in the telephone call. The watchers and/or the automated interaction management system can then use this information as a guide to determine if the person is likely to be able to respond to another interaction while that person is also participating in the telephone call. The inference is that a person who is 'passive participant' is substantially more likely to engage in an IM or even a second phone interaction with another person (e.g., watcher) then if that person was an 'active participant'.

Centrally controlling communication at a venue

One example may include a method that includes receiving, at a server, a data set from one or more mobile devices located in a presentation space, combining the received data set with additional data to create a combined data set, creating a presentation signal based on the combined data set, subtracting a portion of one or more of the data set and the additional data set from the combined data set to create a modified presentation signal, forwarding the modified presentation signal to one or more of a display and a loudspeaker located in the presentation space, and playing the modified presentation signal via one or more of the loudspeaker and the display device.

Communication system and evaluation method

A communication system is configured to broadcast utterance voice data received from one of mobile communication terminals to other mobile communication terminals, to control text delivery such that a result of utterance voice recognition from voice recognition processing on the received utterance voice data is displayed on the mobile communication terminals in synchronization, and to use the result of utterance voice recognition to perform communication evaluation. The communication evaluation includes a first evaluation including evaluating a dialogue between users based on a group dialogue index to produce group communication evaluation information, a second evaluation including evaluating utterances constituting the dialogue between the users based on a personal utterance index to produce personal utterance evaluation information, and a third evaluation including using the group communication evaluation information and the personal utterance evaluation information to produce entire communication group evaluation information.

Shared speakerphone system for multiple devices in a conference room

A speakerphone system is shared with multiple participant devices of participants in a physical meeting that are using a web conferencing service. An active speaker is identified from the participants. The participant device of the active speaker is switched, such that the speakerphone system receives and renders audio of the active speaker. Video of the participant device of the active speaker is enabled, such that the web conferencing service displays the video to the participant devices.

Shared Speakerphone System for Multiple Devices in a Conference Room

A speakerphone system is shared with multiple participant devices of participants in a physical meeting that are using a web conferencing service. An active speaker is identified from the participants. The participant device of the active speaker is switched, such that the speakerphone system receives and renders audio of the active speaker. Video of the participant device of the active speaker is enabled, such that the web conferencing service displays the video to the participant devices.

Ｉｓｍ 대역 다채널 주파수를 이용한 근거리에서의 오디오혼합기능을 갖는 회의용 송수신장치 및 방법

본 발명은 회의실용 전자제품(오디오 기기, 노트북, TV, PC) 일측에 연결되어 복수개의 무선헤드셋을 사용하는 사용자들에게 하나의 무선단말기를 통해 채널별로 선택하여 ISM 대역에서 무선으로 발언내용을 말하고 들을 수 있도록 하기 위해, 상기 무선 단말기는 PWM 전원제어부(PWM POWER & BAT CONTROL), 신호입력부(Signal_Input), 제1 ISM 대역 수신부(RX), 제1 ISM 대역 송신부(TX), 제어부(MCU), 키 패드 제어부로 구성되고, 상기 무선헤드셋은 제2 ISM 대역 수신부(RX), 제2 ISM 대역 송신부(TX), 통화제어부로 이루어진 회의용 송수신 장치가 구성됨으로서, 하나의 회의용 단말기를 통해 복수개의 무선헤드셋 중 특정 무선헤드셋에서 전송되는 발언자의 음성신호를 선택하여 에러 및 잡음없이 전송시킬 수 있고, 저 전력으로 송수신할 수 있는 ISM 대역 다채널 주파수를 이용한 근거리에서의 오디오 혼합기능을 갖는 회의용 송수신장치 및 방법을 제공하는데 그 목적이 있다. ISM 대역, 회의용 단말기, 무선헤드셋

Conference calls

Example embodiments relate to conference calls. In a method, there may be provided audio data and associated classification data from a plurality of user devices as part of a conference call, the plurality of user devices including at least a first user device and a second user device. The classification data may be indicative of one of a plurality of predetermined sound types represented by the associated audio data in a current time frame. The method may comprise determining that the audio data provided by the first user device is associated with a dominant speaker. The method may comprise controlling transmission of the audio data as part of the conference call, including preventing transmission of audio data provided by at least the second user device based on its associated classification data being indicative of a restricted sound type and the first user device being associated with a dominant speaker.

Hybrid Teleconference platform

Teleconferencing between a plurality of participants (P1-P6) comprises assigning each participant to one of a plurality of clusters (12,14,16), wherein at least one audio capture component and/or at least one audio output component is associated with the clusters and/or participants. An audio stream associated with one of the clusters is captured using at least one of the audio capture components and the audio stream is broadcast to at least one of the other clusters using at least one of the audio output components. The captured audio stream is not broadcast from any of the audio output components associated with the cluster from which the audio stream was captured. A further aspect of the invention provides identifying a single audio capture component from amongst the audio capture components of the audio stream as a primary audio capture component based on an amplitude of the audio streams being within a predetermined frequency range, the range having a lower bound above 2kHz, and broadcasting the primary audio capture component. Another aspect of the invention assigns the participants of the teleconference to clusters indicative of participants physical proximity to one another.

Webcasting mit just-in-time- betriebsmittelbereitstellung, automatisierter signalerfassung und streaming und vollautomatisierter ereignisarchivierung

音频会议装置及音频会议系统

本发明提供一种视频会议系统，其一边在较大范围内接收会议参加者的发言，一边对主要发言者进行拍摄。本发明的音频会议装置(1)使用排列多个传声器(MIC)而构成的传声器阵列，接收较大范围的声音和分割为较小范围的声音。另外，将在较大范围内接收到的音频信号(MB1、MB2)用作声音接收用的音频信号(MB0)。另外，使用根据分成较小范围而接收到的音频信号(MB11～MB14、MB21～MB24)检测出的最高电平的音频信号，检测接收方向(DS)，基于该接收方向(DS)控制照相机(7)的拍摄方向。

Kommunikationseinrichtung zur bereitstellung eines effizienten schlafmodus für ein gruppenkommunikationsnetzwerk

Efficient buffer allocation for current and predicted active speakers in voice conferencing systems

A method and computer program product allows for the efficient allocation of buffers for current and predicted active speakers in voice conferencing systems (200). The method and computer program product, implemented by a server hosting an audio conference for a plurality of speakers, minimizes the loss of audio data for speakers as they switch from "non-active" to "active" status (208). This is accomplished by employing a set of active speaker buffers (210) and a set of predicted active speakers buffers (220). The predicted active speaker buffers (220) maintain a collection of the most recent x packets or milliseconds of "non-active" speaker audio data, and transfer a portion of the data from the predicted active speaker buffers to the active speaker buffers as speakers become "active" speakers. The x packets or m milliseconds of stored "non-active" speaker audio data can be used only up to a pre-determined jitter buffer fill-level in order to avoid introducing additional audio packet delivery delay to participants of the conference.

音声会議システムにおける現行のアクティブな話者及び予測されるアクティブな話者に対する効率的なバッファの割振り

音声会議システム（２００）における現行のアクティブな話者及び予測されるアクティブな話者に対するバッファの効率的な割振りを可能とする方法及びコンピュータプログラム製品。複数の話者に対して音声会議をホストするサーバによって実施されるこの方法及びコンピュータプログラム製品により、話者が「非アクティブな」ステータスから「アクティブな」ステータスに切り換わる際（２０８）、話者に関する音声データの損失が最小限に抑えられる。これは、１組のアクティブな話者バッファ（２１０）、及び１組の予測されるアクティブな話者バッファ（２２０）を使用することによって達せられる。予測されるアクティブな話者バッファ（２２０）は、最新のｘ個のパケットの集合、即ちミリ秒単位の「非アクティブな」話者の音声データを保持し、話者が「アクティブな」話者になる際、そのデータの一部分を予測されるアクティブな話者バッファからアクティブな話者バッファに転送する。このｘ個のパケット、即ちｍミリ秒の記録された「非アクティブな」話者の音声データは、会議の参加者に対する追加の音声パケット送達遅延の導入を回避するために、所定のジッタバッファ充てんレベルまでに限って使用され得る。

Efficient buffer allocation for current and predicted active speakers in voice conferencing systems

A method and computer program product allows for the efficient allocation of buffers for current and predicted active speakers in voice conferencing systems (200). The method and computer program product, implemented by a server hosting an audio conference for a plurality of speakers, minimizes the loss of audio data for speakers as they switch from "non-active" to "active" status (208). This is accomplished by employing a set of active speaker buffers (210) and a set of predicted active speakers buffers (220). The predicted active speaker buffers (220) maintain a collection of the most recent x packets or milliseconds of "non-active" speaker audio data, and transfer a portion of the data from the predicted active speaker buffers to the active speaker buffers as speakers become "active" speakers. The x packets or m milliseconds of stored "non-active" speaker audio data can be used only up to a pre-determined jitter buffer fill-level in order to avoid introducing additional audio packet delivery delay to participants of the conference.

Efficient buffer allocation for current and predicted active speakers in voice conferencing systems

A method and computer program product allows for the efficient allocation of buffers (e.g., first-in first-out (FIFO) queues) for current and predicted active speakers in voice conferencing systems. The method and computer program product, implemented by a server hosting an audio conference for a plurality of speakers, minimizes the loss of audio data for speakers as they switch from “non-active” to “active” status. This is accomplished by employing a set of active speaker buffers and a set of predicted active speaker buffers. The predicted active speaker buffers maintain a collection of the most recent x packets or m milliseconds of “non-active” speaker audio data, and transfer a portion of the data from the predicted active speaker buffers to the active speaker buffers as speakers become “active” speakers. The x packets or m milliseconds of stored “non-active” speaker audio data can be used only up to a pre-determined jitter buffer fill-level in order to avoid introducing additional audio packet delivery delay to participants of the conference.

Effiziente bufferzuteilung fur aktuelle und vorhergesagte aktive sprecher in sprechkonferenzsystemen

音声会議システムにおける現行のアクティブな話者及び予測されるアクティブな話者に対する効率的なバッファの割振り