Method and apparatus for wideband conferencing
    181.
    发明授权
    Method and apparatus for wideband conferencing 有权
    用于宽带会议的方法和装置

    公开(公告)号:US08023458B2

    公开(公告)日:2011-09-20

    申请号:US11679489

    申请日:2007-02-27

    CPC classification number: H04M3/567 H04M11/062

    Abstract: A method and apparatus for wideband voice and optional data conferencing over a telecommunications network channel between at least two wideband communications devices. An exemplary method comprises establishing an audio link, verifying wideband capability between the at least two wideband communications devices, training modems of the at least two wideband communications device to line conditions, and adjusting the telecommunications connection line conditions between the communications devices. Once a wideband connection has been established, audio and data may be simultaneously exchanged.

    Abstract translation: 一种用于在至少两个宽带通信设备之间的电信网络信道上的宽带语音和可选数据会议的方法和装置。 一种示例性方法包括建立音频链路,验证至少两个宽带通信设备之间的宽带能力,将至少两个宽带通信设备的调制解调器训练到线路状况,以及调整通信设备之间的电信连接线路状况。 一旦建立了宽带连接,可以同时交换音频和数据。

    Automatic Audio Priority Designation During Conference
    182.
    发明申请
    Automatic Audio Priority Designation During Conference 有权
    会议期间自动音频优先指定

    公开(公告)号:US20110187814A1

    公开(公告)日:2011-08-04

    申请号:US12697844

    申请日:2010-02-01

    CPC classification number: G10L21/00 H04M3/42 H04N7/15

    Abstract: During a conference, a multipoint control unit (MCU) designates priority and non-priority endpoints. The MCU forms priority audio from the priority endpoint and sends that audio to the other endpoints at a normal level. However, the MCU forms non-priority audio from the non-priority endpoint and handles that audio based on whether the input audio from the priority endpoint is from speaking or not. If the priority endpoint's audio indicates a participant at that endpoint is speaking, then the MCU sends the non-priority audio to the other endpoints at a reduced level. Designation of which endpoint has priority can be based on which endpoint has a current duration of audio indicative of speech that is longer than other endpoints. Alternatively, the designation can be based on which endpoint is currently presenting content during the conference or based on a mix of speech audio and content presentation.

    Abstract translation: 在会议期间,多点控制单元(MCU)指定优先级和非优先级端点。 MCU从优先级端点构成优先音频,并将该音频发送到正常级别的其他端点。 然而,MCU从非优先级终端形成非优先级音频,并且基于来自优先级端点的输入音频是否来自不同语言来处理该音频。 如果优先级端点的音频指示该端点的参与者正在说话,则MCU以降低的级别将非优先级音频发送到其他端点。 哪个端点具有优先权的指定可以基于哪个端点具有指示语音的当前持续时间长于其他端点。 或者,该指定可以基于在会议期间当前呈现内容的哪个端点或者基于语音音频和内容呈现的混合。

    System and method for exchanging connection information for videoconferencing units using instant messaging
    183.
    发明授权
    System and method for exchanging connection information for videoconferencing units using instant messaging 有权
    用于使用即时消息交换视频会议单元的连接信息的系统和方法

    公开(公告)号:US07969461B2

    公开(公告)日:2011-06-28

    申请号:US11277979

    申请日:2006-03-30

    CPC classification number: H04L12/1818 H04L51/04 H04L51/28 H04N7/147 H04N7/15

    Abstract: A videoconferencing system includes a first videoconferencing unit coupled to a network and associated with a first instant messaging identity. The first videoconferencing unit obtains a second instant messaging identity and automatically sends a request instant message requesting videoconferencing connection information to the second instant messaging identity. A second videoconferencing unit is coupled to the network and is associated with the second instant messaging identity. The second videoconferencing unit receives the request instant message and automatically returns a response instant message including videoconferencing connection information to the first instant messaging identity. The first videoconferencing unit receives the response instant message and automatically obtains the videoconferencing connection information from the response instant message. Using the videoconferencing connection information, the first videoconferencing unit initiates a videoconference call with the second videoconference unit.

    Abstract translation: 视频会议系统包括耦合到网络并与第一即时消息身份相关联的第一视频会议单元。 第一视频会议单元获得第二即时消息身份,并且自动地向第二即时消息身份发送请求视频会议连接信息的请求即时消息。 第二视频会议单元耦合到网络并与第二即时消息身份相关联。 第二视频会议单元接收请求即时消息,并将包括视频会议连接信息的响应即时消息自动返回到第一即时消息身份。 第一视频会议单元接收响应即时消息,并从响应即时消息中自动获取视频会议连接信息。 使用视频会议连接信息,第一视频会议单元发起与第二视频会议单元的视频会议呼叫。

    Dual-transform coding of audio signals
    184.
    发明授权
    Dual-transform coding of audio signals 有权
    音频信号的双变换编码

    公开(公告)号:US07953595B2

    公开(公告)日:2011-05-31

    申请号:US11550629

    申请日:2006-10-18

    CPC classification number: G10L19/0212 G10L19/022

    Abstract: Methods, devices, and systems for coding and decoding audio are disclosed. At least two transforms are applied on an audio signal, each with different transform periods for better resolutions at both low and high frequencies. The transform coefficients are selected and combined such that the data rate remains similar as a single transform. The transform coefficients may be coded with a fast lattice vector quantizer. The quantizer has a high rate quantizer and a low rate quantizer. The high rate quantizer includes a scheme to truncate the lattice. The low rate quantizer includes a table based searching method. The low rate quantizer may also include a table based indexing scheme. The high rate quantizer may further include Huffman coding for the quantization indices of transform coefficients to improve the quantizing/coding efficiency.

    Abstract translation: 公开了用于编码和解码音频的方法,设备和系统。 在音频信号上应用至少两个变换,每个变换具有不同的变换周期,以便在低频和高频都能获得更好的分辨率。 选择和组合变换系数,使得数据速率与单个变换保持相似。 可以用快速格子矢量量化器对变换系数进行编码。 量化器具有高速率量化器和低速率量化器。 高速率量化器包括截短晶格的方案。 低速率量化器包括基于表的搜索方法。 低速率量化器还可以包括基于表的索引方案。 高速率量化器还可以包括用于变换系数的量化索引的霍夫曼编码,以提高量化/编码效率。

    Method and system for handling video signals of conference
    185.
    发明授权
    Method and system for handling video signals of conference 有权
    会议视频信号处理方法及系统

    公开(公告)号:US07929011B2

    公开(公告)日:2011-04-19

    申请号:US11148103

    申请日:2005-06-08

    CPC classification number: H04N7/15

    Abstract: A system and method particularly useful in handling the video of a “large” conference is disclosed. The system and method initially defines certain conferees as panel members, which can provide input video signals to the multipoint control unit and can receive output video signals or a composite layout for the conference from the multipoint control unit. The system and method also initially defines audience members, which only passively receive the output video signals or layout from the multipoint control unit and do not have their video signals incorporated into the layout of the videoconference. In one aspect, an audience terminal can request to participate in the conference. By making an appropriate request, the audience terminal can, if system resources are adequate, participate fully as a panel member and may supply video signals to the conference for at least some portion of time. Alternatively, the audience member can supply information other than video signals to the conference, such as voting data or other data allowing the audience member to participate in the conference as more than a merely passive viewing participant.

    Abstract translation: 公开了一种特别适用于处理“大”会议视频的系统和方法。 系统和方法最初将某些与会者定义为面板成员,其可以向多点控制单元提供输入视频信号,并且可以从多点控制单元接收输出视频信号或会议的复合布局。 该系统和方法还最初定义观众成员,其仅被动地接收来自多点控制单元的输出视频信号或布局,并且其视频信号不包括在视频会议的布局中。 一方面,观众终端可以请求参加会议。 通过提出适当的要求,如果系统资源充足,观众终端可以作为小组成员充分参与,并且可以至少在一段时间内向会议提供视频信号。 或者,观众可以向会议提供除了视频信号之外的信息,例如投票数据或允许观众参与会议的其他数据,不仅仅是被动观看参与者。

    Detection and Suppression of Returned Audio at Near-End
    186.
    发明申请
    Detection and Suppression of Returned Audio at Near-End 有权
    返回音频在近端的检测和抑制

    公开(公告)号:US20110069830A1

    公开(公告)日:2011-03-24

    申请号:US12565374

    申请日:2009-09-23

    CPC classification number: H04M3/002 H04M1/6033 H04M3/2236 H04M3/568 H04M9/085

    Abstract: Audio from a near-end that has been acoustically coupled at the far-end and returned to the near-end unit is detected and suppressed at the near-end of a conference. First and second energy outputs for separate bands are determined for the near-end audio being sent from the near-end unit and for the far-end audio being received at the near-end unit. The near-end unit compares the first and second energy outputs to one another for each of the bands over a time delay range and detects the return of the sent near-end audio in the received far-end audio based on the comparison. The comparison can use a cross-correlation to find an estimated time delay used for further analysis of the near and far-end energies. The near-end unit suppresses any detected return by muting or reducing what far-end audio is output at its loudspeaker.

    Abstract translation: 来自远端的声耦合并返回到近端单元的近端的音频在会议的近端被检测和抑制。 确定用于单独频带的第一和第二能量输出用于从近端单元发送的近端音频以及在近端单元处接收的远端音频。 近端单元在时间延迟范围内针对每个频带将第一和第二能量输出彼此进行比较,并且基于该比较来检测所接收的远端音频中所发送的近端音频的返回。 比较可以使用互相关来找到用于进一步分析近端和远端能量的估计时间延迟。 近端单元通过静音抑制任何检测到的返回,或者减少在其扬声器处输出远端音频。

    Remote multipoint architecture for full-duplex audio
    187.
    发明授权
    Remote multipoint architecture for full-duplex audio 有权
    用于全双工音频的远程多点架构

    公开(公告)号:US07903828B2

    公开(公告)日:2011-03-08

    申请号:US11065792

    申请日:2005-02-25

    CPC classification number: H04M9/082

    Abstract: A method, microphone module and a system for full-duplex audio system implementing full duplex audio, audio echo cancellation and audio codec. The audio signals from the microphones and to the loudspeakers are encoded to reduce the transmission bandwidth requirement during wired or wireless transmission and decoded when needed for further processing. Each microphone has transceivers to receive signals fed to loudspeakers within the microphone module. An audio echo canceller is installed within each microphone module to generate echo free audio signal for further signal processing. Problems due to the combination of lossy codec and echo cancellation are avoided.

    Abstract translation: 一种方法,麦克风模块和全双工音频系统实现全双工音频,音频回声消除和音频编解码器的系统。 编码麦克风和扬声器的音频信号以减少有线或无线传输期间的传输带宽需求,并在需要进一步处理时进行解码。 每个麦克风具有收发器以接收馈送到麦克风模块内的扬声器的信号。 音频回声消除器安装在每个麦克风模块内,以产生无回声的音频信号用于进一步的信号处理。 避免了有损编解码器与回波消除相结合的问题。

    DIGITAL LINKING OF MULTIPLE MICROPHONE SYSTEMS
    188.
    发明申请
    DIGITAL LINKING OF MULTIPLE MICROPHONE SYSTEMS 有权
    数字麦克风系统的数字连接

    公开(公告)号:US20100278358A1

    公开(公告)日:2010-11-04

    申请号:US12838215

    申请日:2010-07-16

    CPC classification number: H04M3/562 H04M3/561 H04M3/568 H04R3/005

    Abstract: Systems and methods for digitally linking multiple microphones and managing microphone signals are provided. Embodiments provide for digitally linking of multiple auto-mixer systems where a large number of microphones are required. In large auto-mixer systems, microphone input channels may be distributed among many devices. Rather than transmitting these signals individually to a central processing device for the auto-mixer, it is most efficient to distribute the audio signal processing functions of the automatic microphone mixer among multiple devices in an audio conferencing system. Subsequently, a smaller number of gated mixes of microphone signals may be transmitted between devices. Thus, embodiments of the present invention act to digitally link multiple microphone signals, arbitrating these signals, in order to enable distributed automatic microphone mixers to behave as a single mixer.

    Abstract translation: 提供了用于数字连接多个麦克风和管理麦克风信号的系统和方法。 实施例提供了需要大量麦克风的多个自动混合器系统的数字连接。 在大型自动混音器系统中,麦克风输入通道可以分布在许多设备中。 将这些信号单独地发送到用于自动混频器的中央处理设备,而不是在音频会议系统中的多个设备之间分配自动麦克风混频器的音频信号处理功能是最有效的。 随后,可以在设备之间传输较少数量的麦克风信号的门控混合。 因此,本发明的实施例用于数字地链接多个麦克风信号,仲裁这些信号,以使分布式自动麦克风混频器能够作为单个混频器。

    System and method for improving the quality of video communication over a packet-based network
    189.
    发明授权
    System and method for improving the quality of video communication over a packet-based network 有权
    用于提高基于分组的网络的视频通信质量的系统和方法

    公开(公告)号:US07818772B2

    公开(公告)日:2010-10-19

    申请号:US10199872

    申请日:2002-07-18

    Abstract: A system and method for improving quality of video communication includes a packets generator logical unit (PGU) coupled to a video source for receiving a compressed video stream and generating primary and secondary data chunks for transmittal to one or more destination nodes via a network. Each primary data chunk is associated with a secondary data chunk, and each secondary data chunk includes video data that covers a same portion of a video frame as video data included in each associated primary data chunk. The PGU encapsulates the primary and secondary data chunks in associated primary-secondary packet pairs composed of MB data fields, picture header fields, GOB and slice start header fields, and MB header fields, among others. The destination nodes process the secondary packets received via the network if the primary packets are damaged or lost in transmission.

    Abstract translation: 用于提高视频通信质量的系统和方法包括耦合到视频源的分组生成器逻辑单元(PGU),用于接收压缩视频流,并且经由网络生成用于传送到一个或多个目的地节点的主和辅助数据块。 每个主要数据块与辅助数据块相关联,并且每个辅助数据块包括覆盖视频帧的与视频数据相同部分的视频数据,视频数据块包括在每个关联的主数据块中。 PGU将主数据块和辅助数据块封装在由MB数据字段,图片头字段,GOB和片起始头字段以及MB头字段组成的关联的主二次分组中。 如果主要分组在传输中损坏或丢失,则目的节点处理经由网络接收到的二次分组。

    Method for determining DFS channel availability in a wireless LAN
    190.
    发明授权
    Method for determining DFS channel availability in a wireless LAN 有权
    确定无线LAN中DFS信道可用性的方法

    公开(公告)号:US07813744B2

    公开(公告)日:2010-10-12

    申请号:US11513763

    申请日:2006-08-31

    Inventor: Paul D. Johnson

    CPC classification number: H04W36/30

    Abstract: A LAN has a wireless portion which includes access points and wireless communications devices and a wired portion which includes such things as routers, servers, and a PBX. The LAN operates in a radio frequency band where certain channels (DFS channels) are shared with a signal that is of higher priority than the communications signals transmitted and received by the access point and the wireless communications devices. If a higher priority signal is sensed on a particular channel of the wireless medium by a master device, it is necessary for all of the slave devices communicating on this channel to stop transmitting on that channel within a short period of time. The wireless communications devices are mobile and can be easily transported around in space or roam during a communications session. While roaming during a communications session, it is likely that the communications device will move out of range of an access point with which it is currently associated and possibly cause the session to end prematurely, and so it is important that a handoff procedure operates so that a communications session is continuous as the communications device roams from access point to access point. By setting and maintaining a timer dedicated to each available DFS channel, it is possible to ensure that the communications sessions is not interrupted during the roaming process in an environment were certain channels are shared with a higher priority signal.

    Abstract translation: LAN具有包括接入点和无线通信设备的无线部分以及包括诸如路由器,服务器和集团电话之类的有线部分。 LAN工作在无线电频段,其中某些信道(DFS信道)与比由接入点和无线通信设备发送和接收的通信信号优先的信号共享。 如果由主设备在无线介质的特定信道上感测到较高优​​先级信号,则在该信道上通信的所有从设备在短时间段内停止在该信道上的传输是必要的。 无线通信设备是移动的,并且可以在通信会话期间容易地在空间中漫游或漫游。 在通信会话期间漫游时,通信设备很可能移动超出其当前关联的接入点的范围,并且可能导致会话过早结束,因此重要的是切换过程操作使得 当通信设备从接入点漫游到接入点时,通信会话是连续的。 通过设置和维护专用于每个可用的DFS信道的定时器,可以确保通信会话在环境中的漫游过程期间不被中断,某些信道与较高优先级信号共享。

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