会员体验
专利管家(专利管理)
工作空间(专利管理)
风险监控(情报监控)
数据分析(专利分析)
侵权分析(诉讼无效)
联系我们
交流群
官方交流:
QQ群: 891211   
微信请扫码    >>>
现在联系顾问~
热词
    • 4. 发明授权
    • Architecture for an extensible real-time collaboration system
    • 可扩展实时协作系统架构
    • US08321506B2
    • 2012-11-27
    • US10918855
    • 2004-08-14
    • Mu HanKrishnamurthy GanesanAdrian PotraNikhil Bobde
    • Mu HanKrishnamurthy GanesanAdrian PotraNikhil Bobde
    • G06F15/16H04Q11/00
    • H04L65/1006G06F9/54H04L29/06H04L29/06027H04L65/403
    • An architecture for an extensible real-time collaboration system is provided. The architecture presents a unified application program interface for writing application programs that use communications protocols. The architecture has activity objects, endpoint objects, and multiple media stacks. These objects may use various communications protocols, such as Session Initiation Protocol or Real-Time Transport Protocol to send and receive messages. The activity objects, endpoint objects, and multiple media stacks may each have one or more APIs that an application developer can use to access or provide collaboration-related functionality. These objects map the API to the underlying implementation provided by other objects. Using the activity objects enables a developer to provide less application logic than would otherwise be necessary to provide complex collaboration services.
    • 提供了可扩展实时协作系统的架构。 该架构提供了一个统一的应用程序界面,用于编写使用通信协议的应用程序。 该架构具有活动对象,端点对象和多个媒体堆栈。 这些对象可以使用各种通信协议,例如会话发起协议或实时传输协议来发送和接收消息。 活动对象,端点对象和多个媒体堆栈可以各自具有应用开发者可以用来访问或提供协作相关功能的一个或多个API。 这些对象将API映射到其他对象提供的底层实现。 使用活动对象使开发人员能够提供比提供复杂协作服务所必需的更少的应用程序逻辑。
    • 6. 发明授权
    • Selective glitch detection, clock drift compensation, and anti-clipping in audio echo cancellation
    • 选择性毛刺检测,时钟漂移补偿和音频回声消除中的抗剪辑
    • US08295475B2
    • 2012-10-23
    • US11332500
    • 2006-01-13
    • Qin LiChao HeWei-Ge ChenMu Han
    • Qin LiChao HeWei-Ge ChenMu Han
    • H04M9/08
    • H04M9/082
    • The quality and robustness of audio echo cancellation is enhanced by selectively applying glitch recovery processes based on a quality measurement of the relative offset between capture and render audio streams. For example, large and small glitch detection is enabled for low relative offset variance; large glitch detection is enabled in a medium range of relative offset variance; and neither enabled at high variance. Further, a fast glitch recovery process suspends updating the adaptive filter coefficients of the audio echo cancellation while buffers are re-aligned to recover from the glitch, so as to avoid resetting the adaptive filter. When clock drift exists between capture and render audio streams, a multi-step compensation method is applied to improve AEC output quality in case the drifting rate is low; and a resampler is used to compensate the drift in case the drifting rate is high. An anti-clipping process detects clipping of the signals, and also suspends adaptive filter updating during clipping.
    • 通过基于捕获和渲染音频流之间的相对偏移的质量测量来选择性地应用毛刺恢复过程来增强音频回声消除的质量和鲁棒性。 例如,对于低相对偏移方差,启用大的和小的毛刺检测; 在相对偏移方差的中等范围内启用大毛刺检测; 并且在高方差时都不启用。 此外,快速毛刺恢复处理暂停更新音频回声消除的自适应滤波器系数,同时缓冲器被重新对准以从毛刺恢复,以避免复位自适应滤波器。 当捕获和渲染音频流之间存在时钟漂移时,应用多步补偿方法来提高漂移率低的AEC输出质量; 并且在漂移速率高的情况下使用重采样器来补偿漂移。 反剪辑过程检测信号的剪辑,并且还可以在剪辑期间暂停自适应滤波器更新。
    • 8. 发明授权
    • Method and system for providing adaptive bandwidth control for real-time communication
    • 为实时通信提供自适应带宽控制的方法和系统
    • US07554922B2
    • 2009-06-30
    • US11560445
    • 2006-11-16
    • Andres Vega-GarciaMu HanQianbo Huai
    • Andres Vega-GarciaMu HanQianbo Huai
    • H04J3/14
    • H04L65/608H04L29/06H04L29/06027H04L47/10H04L47/115H04L47/263H04L47/283H04L65/80
    • A method and system for dynamically altering the transmission settings of one or more computing devices engaged in a real-time communication session is presented. The devices exchange meaningful and dummy control packets according to a standard control protocol. The approximate bandwidth available on the network is then calculated based on the difference in arrival times between at least one of the dummy control packets and at least one of the meaningful control packets. Once the approximate bandwidth available on the network is computed, the one or more devices adjust outgoing audio and video data streams using a quality control mechanism. The quality control mechanism enables the one or more devices to transmit data in a way that maximizes the user experience during the real-time communication session.
    • 提出了一种用于动态地改变参与实时通信会话的一个或多个计算设备的传输设置的方法和系统。 设备根据标准控制协议交换有意义的和虚拟的控制数据包。 然后基于至少一个虚拟控制分组与至少一个有意义的控制分组之间的到达时间的差异来计算网络上可用的大致带宽。 一旦计算了网络上可用的大致带宽,则一个或多个设备使用质量控制机制来调整输出的音频和视频数据流。 质量控制机制使得一个或多个设备能够以在实时通信会话期间最大化用户体验的方式来发送数据。
    • 9. 发明申请
    • REDUCING INFORMATION RECEPTION DELAYS
    • 减少信息接收延迟
    • US20080294793A1
    • 2008-11-27
    • US11951912
    • 2007-12-06
    • Mu HanAndres Vega GarciaWei Zhong
    • Mu HanAndres Vega GarciaWei Zhong
    • G06F15/16
    • H04L12/6418H04L2012/6472Y10S345/951
    • A technique for reducing information reception delays is provided. The technique reduces delays that may be caused by protocols that guarantee order and delivery, such as TCP/IP. The technique creates multiple connections between a sender and recipient computing devices and sends messages from the sender to the recipient on the multiple corrections redundantly. The recipient can then use the first arriving message and ignore the subsequently arriving redundant messages. The recipient can also wait for a period of time before determining which of the arrived messages to use. The technique may dynamically add connections if messages are not consistently received in a timely manner on multiple connections. Conversely, the technique may remove connections if messages are consistently received in a timely manner on multiple connections. The technique can accordingly be used with applications that are intolerant of data reception delays such as Voice over IP, real-time streaming audio, or real-time streaming video.
    • 提供了用于减少信息接收延迟的技术。 该技术减少了可能由保证订单和传递的协议(如TCP / IP)引起的延迟。 该技术在发送方和收件人计算设备之间创建多个连接,并以多次更正方式从发送方向接收方发送消息。 接收者可以使用第一个到达的消息,并忽略随后到达的冗余消息。 收件人还可以等待一段时间才能确定要使用的到达消息。 如果在多个连接上不及时地接收到消息,则该技术可以动态地添加连接。 相反,如果在多个连接上一致地接收到消息,则该技术可以去除连接。 因此,该技术可以与不耐受诸如IP语音,实时流音频或实时流视频之类的数据接收延迟的应用一起使用。
    • 10. 发明授权
    • Multipoint processing unit
    • 多点处理单元
    • US07698365B2
    • 2010-04-13
    • US11838798
    • 2007-08-14
    • Michael R. Van BuskirkPhilippe FerriereMu Han
    • Michael R. Van BuskirkPhilippe FerriereMu Han
    • G06F15/16G06F15/173
    • H04L65/4038H04L12/1822H04L65/103H04L65/104H04L65/605H04L69/08H04N7/152
    • A system to provide a multipoint processing terminal and a multicast bridging terminal to provide mixing, switching, and other processing of media streams under the control of H.323 components. Application Programming Interfaces defined for the multipoint processing terminal provide a multipoint control unit with the capability to change the default behavior of the multipoint processing terminal by allowing the multipoint control unit to control the routing audio and video streams in the multipoint processing terminal and control the media formats in a multipoint conference. Multipoint processing acceleration functionality is provided by providing interfaces to allow hardware accelerated implementations of multipoint processing terminals. The multicast bridging terminals enables clients using one type of control signaling and media streaming to join other conferences using different types of control signaling and media streaming by receiving audio or video data from an incoming media stream and performing any processing necessary to transform the media stream from the incoming stream data format to the outgoing stream data format.
    • 一种提供多点处理终端和组播桥接终端的系统,用于在H.323组件的控制下提供媒体流的混合,切换和其他处理。 应用程序编程为多点处理终端定义的接口提供多点控制单元,具有通过允许多点控制单元控制多点处理终端中的路由音频和视频流并控制媒体的能力来改变多点处理终端的默认行为的能力 多点会议中的格式。 通过提供允许多点处理终端的硬件加速实现的接口来提供多点处理加速功能。 组播桥接终端使得客户端能够使用一种类型的控制信令和媒体流来使用不同类型的控制信令和媒体流来连接其他会议,通过从传入的媒体流接收音频或视频数据,并执行将媒体流从 输入流数据格式为输出流数据格式。