METHOD FOR IMPROVING PERCEPTUAL CONTINUITY IN A SPATIAL TELECONFERENCING SYSTEM
    2.
    发明申请
    METHOD FOR IMPROVING PERCEPTUAL CONTINUITY IN A SPATIAL TELECONFERENCING SYSTEM 审中-公开
    在空间电信系统中改善连续性的方法

    公开(公告)号:WO2014052431A1

    公开(公告)日:2014-04-03

    申请号:PCT/US2013/061648

    申请日:2013-09-25

    CPC classification number: H04M3/561 H04M3/569 H04M2203/5072

    Abstract: 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).

    Abstract translation: 本文件涉及音频会议系统。 特别地,本文件涉及改善音频会议系统内的感知连续性。 根据一方面,描述了用于复用第一和第二连续输入音频信号的方法,以产生将被呈现给收听者的多路复用输出音频信号。 第一和第二输入音频信号(123)分别指示由第一和第二端点(120,170)捕获的声音。 该方法包括分别在第一和第二输入音频信号(123)中确定通话活动(201,202); 以及基于所述第一和/或第二输入音频信号(123)并经受一个或多个复用条件来确定所述复用的输出音频信号。 所述一个或多个复用条件包括:在时刻,当所述第一输入音频信号(123)中存在通话活动(201)时,至少基于所述第一输入音频信号(123)确定所述多路复用的输出音频信号; 在第二输入音频信号(123)中存在通话活动(202)的情况下,至少基于第二输入音频信号(123)确定多路复用的输出音频信号; 并且在静音时刻,当在第一和第二输入音频信号(123)中没有通话活动(201,202)时,仅基于第一和第二输入音频信号中的一个确定多路复用的输出音频信号 (123)。

    LONG TERM MONITORING OF TRANSMISSION AND VOICE ACTIVITY PATTERNS FOR REGULATING GAIN CONTROL
    3.
    发明申请
    LONG TERM MONITORING OF TRANSMISSION AND VOICE ACTIVITY PATTERNS FOR REGULATING GAIN CONTROL 审中-公开
    用于调节增益控制的传输和语音活动模式的长期监测

    公开(公告)号:WO2014043024A1

    公开(公告)日:2014-03-20

    申请号:PCT/US2013/058735

    申请日:2013-09-09

    Abstract: The present document relates to audio communication systems. In particular, the present document relates to the control of the level of audio signals within audio communication systems. A method for leveling a near-end audio signal (211) using a leveling gain (214) is described. The near-end audio signal (211) comprises a sequence of segments, wherein the sequence of segments comprises a current segment and one or more preceding segments. The method comprises determining a nuisance measure (416) which is indicative of an amount of aberrant voice activity within the sequence of segments of the near-end audio signal (211); and determining the leveling gain (214) for the current segment of the near-end audio signal (211), at least based on the leveling gain (214) for the one or more preceding segments of the near-end audio signal (211), and by taking into account - according to a variable degree - an estimate of the level of the current segment of the near-end audio signal (211); wherein the variable degree is dependent on the nuisance measure (416).

    Abstract translation: 本文件涉及音频通信系统。 特别地,本文件涉及音频通信系统中的音频信号的级别的控制。 描述了使用调平增益(214)来调平近端音频信号(211)的方法。 近端音频信号(211)包括段序列,其中片段序列包括当前片段和一个或多个先前片段。 该方法包括确定指示近端音频信号(211)的段的序列内的异常语音活动量的扰动度量(416); 以及至少基于近端音频信号(211)的一个或多个先前段的调平增益(214)确定近端音频信号(211)的当前段的调平增益(214) 并且根据可变程度考虑近端音频信号(211)的当前段的电平的估计; 其中所述可变度取决于所述妨扰措施(416)。

    VOICE COMMUNICATION METHOD AND APPARATUS AND METHOD AND APPARATUS FOR OPERATING JITTER BUFFER
    5.
    发明申请
    VOICE COMMUNICATION METHOD AND APPARATUS AND METHOD AND APPARATUS FOR OPERATING JITTER BUFFER 审中-公开
    语音通信方法和装置以及操作抖动缓冲器的方法和装置

    公开(公告)号:WO2013142705A1

    公开(公告)日:2013-09-26

    申请号:PCT/US2013/033332

    申请日:2013-03-21

    CPC classification number: H04L49/90 G10L19/167 G10L25/78 H04L65/1066 H04M3/569

    Abstract: Voice communication method and apparatus and method and apparatus for operating jitter buffer are described. Audio blocks are acquired in sequence. Each of the audio blocks includes one or more audio frames. Voice activity detection is performed on the audio blocks. In response to deciding voice onset for a present one of the audio blocks, a subsequence of the sequence of the acquired audio blocks is retrieved. The subsequence precedes the present audio block immediately. The subsequence has a predetermined length and non-voice is decided for each audio block in the subsequence. The present audio block and the audio blocks in the subsequence are transmitted to a receiving party. The audio blocks in the subsequence are identified as reprocessed audio blocks. In response to deciding non-voice for the present audio block, the present audio block is cached.

    Abstract translation: 描述了用于操作抖动缓冲器的语音通信方法和装置及方法和装置。 按顺序获取音频块。 每个音频块包括一个或多个音频帧。 对音频块执行语音活动检测。 响应于为音频块中的当前音频块确定语音开始,检索获取的音频块的序列的子序列。 该子序列立即在当前音频块之前。 子序列具有预定长度,并且在子序列中为每个音频块确定非声音。 当前音频块和子序列中的音频块被发送到接收方。 子序列中的音频块被识别为再处理的音频块。 响应于为当前音频块确定非语音,缓存当前音频块。

    MULTIPOINT CONNECTION APPARATUS AND COMMUNICATION SYSTEM
    6.
    发明申请
    MULTIPOINT CONNECTION APPARATUS AND COMMUNICATION SYSTEM 审中-公开
    多点连接装置和通信系统

    公开(公告)号:WO2013008941A1

    公开(公告)日:2013-01-17

    申请号:PCT/JP2012/068039

    申请日:2012-07-10

    Inventor: AIBA, Akihito

    Abstract: A multipoint connection apparatus (200) includes a video/audio-signal receiving unit (201) that receives video/audio signals from video/audio terminals (100); a volume-level calculating unit (205) that calculates volume levels from the video/audio signals; a volume-display-image generating unit (207) that generates volume display images indicating volume from the volume levels; a layout-setting-information receiving unit (209) that receives layout setting information indicating information about arrangement of videos to be displayed on the video/audio terminal (100); a combined-video/audio-signal generating unit (211) that generates a combined video/audio signal by combining the video/audio signals and the volume display images based on the layout setting information; and a transmitting unit (215) that transmits the video/audio signal to the video/audio terminal (100).

    Abstract translation: 多点连接装置(200)包括从视频/音频终端(100)接收视频/音频信号的视频/音频信号接收单元(201)。 音量电平计算单元,用于根据视频/音频信号计算音量; 体积显示图像生成单元,其从体积级生成表示体积的体积显示图像; 布局设置信息接收单元(209),其接收关于要在所述视频/音频终端(100)上显示的视频的布置的信息的布局设置信息; 组合视频/音频信号产生单元,其通过基于布局设置信息组合视频/音频信号和音量显示图像来产生组合的视频/音频信号; 以及将视频/音频信号发送到视频/音频终端(100)的发送单元(215)。

    MANAGING A PACKET SWITCHED CONFERENCE CALL
    8.
    发明申请
    MANAGING A PACKET SWITCHED CONFERENCE CALL 审中-公开
    管理一个分组交换会议呼叫

    公开(公告)号:WO2004006475A3

    公开(公告)日:2005-04-14

    申请号:PCT/IB0202625

    申请日:2002-07-04

    Abstract: The invention relates to a method for managing a packet switched, centralized conference call between a plurality of terminals (13). In order to enable an enhancement of the user comfort, it is proposed that the method comprises at a conference call server (12) receiving data packets from all terminals (13). Based on these data packets, then at least one terminal (13) currently providing voice data is determined. In a next step, the data received in the data packets is mixed, and the mixed data is inserted into new data packets together with at least one identifier associated to one of the terminals (13) which were determined to provide voice data, such that the at least one identifier can be distinguished from any other information in the data packets. Finally, the new data packets are transmitted to terminals (13) participating in the conference call. The invention relates equally to a corresponding server and to a corresponding terminal.

    Abstract translation: 本发明涉及一种用于在多个终端(13)之间管理分组交换的集中式电话会议的方法。 为了能够提高用户舒适性,提出该方法包括在会议呼叫服务器(12)处接收来自所有终端(13)的数据分组。 基于这些数据分组,确定当前提供语音数据的至少一个终端(13)。 在下一步骤中,将数据分组中接收的数据混合,并将混合数据与至少一个与被确定提供语音数据的终端(13)相关联的标识符一起被插入到新数据分组中,使得 可以将至少一个标识符与数据分组中的任何其他信息区分开。 最后,将新的数据包发送到参与电话会议的终端(13)。 本发明同样涉及对应的服务器和对应的终端。

    METHOD AND APPARATUS FOR IMPROVING LISTENER DIFFERENTIATION OF TALKERS DURING A CONFERENCE CALL
    9.
    发明申请
    METHOD AND APPARATUS FOR IMPROVING LISTENER DIFFERENTIATION OF TALKERS DURING A CONFERENCE CALL 审中-公开
    用于在会议召唤期间改善听众的听众差异的方法和装置

    公开(公告)号:WO2004010414A1

    公开(公告)日:2004-01-29

    申请号:PCT/US2003/022354

    申请日:2003-07-17

    Abstract: A method and associated apparatus for indicating the voice of each talker from a plurality of talkers to be heard by a listener. A talker indicator (Fig. 2, 32) is provided proximate to the listener. Talker identification information is generated in the talker indicator that can be used to indicate the identity of each talker who is speaking at any given time to the listener. A device (Fig. 1, 23) is coupled to the talker indicator that can transmit the voice signal from each talker to the listener. In different aspects, the talker identification information can include such varied indicators as audio, video, or an announcement combined with a temporally compressed voice signal. In another aspect, an emotographic figure is displayed to the listener that each represent a distinct talker (Fig. 12). The mood of each emotographic is somehow configured to reflect the mode of the talker, as indicated by the talker's voice (Fig. 14).

    Abstract translation: 一种用于从听众听到的多个讲话者中指示每个说话人的声音的方法和相关联的装置。 在收听者附近提供讲话人指示符(图2,32)。 讲话者识别信息在讲话者指示符中产生,可用于指示在任何给定时间向听众发言的每个讲话者的身份。 一个设备(图1,23)被耦合到可以将语音信号从每个讲话者发送到收听者的讲话者指示符。 在不同方面,讲话人识别信息可以包括诸如音频,视频或与时间压缩的语音信号组合的通知之类的变化的指示符。 在另一方面,向收听者显示每个表示不同说话者的图形图(图12)。 每个emotographic的心情以某种方式被配置为反映说话者的模式,如讲者的声音所示(图14)。

    PACKET-BASED CONFERENCING
    10.
    发明申请
    PACKET-BASED CONFERENCING 审中-公开
    基于分组的会议

    公开(公告)号:WO0225908A3

    公开(公告)日:2003-04-10

    申请号:PCT/CA0101298

    申请日:2001-09-13

    Abstract: Numerous packet-based terminals coupled within a packet-based network can establish a voice conference without the use of a conference bridge if the packet-based terminals can support specific operations. These specific operations include receiving voice data packets from each of the other packet-based terminals within the voice conference, determining a set of talkers within the voice conference and processing the received media data packets appropriately for the selected set of talkers so as to output uncompressed voice signals corresponding to the talkers to a speaker coupled to the packet-based terminal. the removal of the conference bridge can allow the packet-based apparatus to become independent from the packet-based network administration. Further, the removal of the conference bridge allows a reduction in transcoding and hence, allows a better quality signal to be received at the individual apparatus.

    Abstract translation: 如果基于分组的终端可以支持特定的操作,则在基于分组的网络中耦合的许多基于分组的终端可以建立语音会议而不使用会议桥。 这些具体操作包括从语音会议内的每个其他基于分组的终端接收语音数据分组,确定语音会议内的一组讲话者,并适当地为所选择的一组讲话者处理所接收的媒体数据分组,以输出未压缩 与讲话者对应的话音信号连接到与基于分组的终端相连的扬声器。 会议桥的移除可以允许基于分组的设备独立于基于分组的网络管理。 此外,去除会议桥允许减少代码转换,因此允许在单独的装置处接收更好的质量信号。

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