CONFERENCE SYSTEM AND PROCESS FOR VOICE ACTIVATION IN THE CONFERENCE SYSTEM
    1.
    发明授权
    CONFERENCE SYSTEM AND PROCESS FOR VOICE ACTIVATION IN THE CONFERENCE SYSTEM 有权
    会议系统及方法语音激活这次会议系统

    公开(公告)号:EP2939407B1

    公开(公告)日:2017-02-22

    申请号:EP12816078.5

    申请日:2012-12-27

    申请人: Robert Bosch GmbH

    IPC分类号: H04M3/56

    摘要: 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).

    EFFICIENT BUFFER ALLOCATION FOR CURRENT AND PREDICTED ACTIVE SPEAKERS IN VOICE CONFERENCING SYSTEMS
    4.
    发明授权
    EFFICIENT BUFFER ALLOCATION FOR CURRENT AND PREDICTED ACTIVE SPEAKERS IN VOICE CONFERENCING SYSTEMS 有权
    对于当前的高效缓冲区分配和预测有源音箱系统会议上的演讲

    公开(公告)号:EP1391103B1

    公开(公告)日:2009-03-25

    申请号:EP02703219.2

    申请日:2002-01-25

    发明人: KWAN, Katherine

    摘要: 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.

    Network mute feature in wireless telecommunications systems
    7.
    发明公开
    Network mute feature in wireless telecommunications systems 有权
    在无线通信系统网络静音功能

    公开(公告)号:EP0989766A3

    公开(公告)日:2000-04-19

    申请号:EP99307301.4

    申请日:1999-09-14

    IPC分类号: H04Q7/22 H04Q7/38

    摘要: 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

    公开(公告)号:EP2901668B1

    公开(公告)日:2018-11-14

    申请号:EP13779436.8

    申请日:2013-09-25

    IPC分类号: H04M3/56

    摘要: 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).

    CONFERENCE SYSTEM AND PROCESS FOR VOICE ACTIVATION IN THE CONFERENCE SYSTEM
    9.
    发明公开

    公开(公告)号:EP2939407A1

    公开(公告)日:2015-11-04

    申请号:EP12816078.5

    申请日:2012-12-27

    申请人: Robert Bosch GmbH

    IPC分类号: H04M3/56

    摘要: 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).

    摘要翻译: 会议系统例如在讨论中使用,并且通常包括具有麦克风的多个代表单元,由此在讨论中每个讨论参与者使用他自己的委托单元。 通常,代表单元具有开关等,允许代表单元前面的参与者请求麦克风被激活,使得参与者的语音被输入会议系统并被会议系统放大。 一种会议系统,包括(1)多个代表单元(2),每个代表单元(2)具有麦克风(5),用于从周围接收音频信号;中央服务模块(3),处理多个贡献通道 ,由此提出贡献通道的音频输出对会议系统(1)的放大的音频输出有贡献,由此每个代表单元(2)适于向中央服务模块(3)发送贡献通道提交的请求 ),由此中央服务模块(2)适于授权该请求并向请求委托单元(2)分配贡献信道,从而将请求委托单元i设置为活动状态A,由此委托单元(2 )适于通过语音激活来触发请求,由此在满足至少第一触发条件的情况下触发该请求,从而将代表单元(2)中的一个的音频信号电平定义为可能的请求委托u 对于处于活动状态A的每个其他委托单元(2),i i i高于单独的测试值,由此单独的测试值是由提供给该测试值的音频或语音信号产生的可能请求单元i的估计音频信号电平 其他活动代表单位(2)。

    METHOD FOR IMPROVING PERCEPTUAL CONTINUITY IN A SPATIAL TELECONFERENCING SYSTEM
    10.
    发明公开
    METHOD FOR IMPROVING PERCEPTUAL CONTINUITY IN A SPATIAL TELECONFERENCING SYSTEM 审中-公开
    方法提高感知的连续性,一个空间会议系统TELE的

    公开(公告)号:EP2901668A1

    公开(公告)日:2015-08-05

    申请号:EP13779436.8

    申请日:2013-09-25

    IPC分类号: H04M3/56

    摘要: 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).