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    • 4. 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。
    • 6. 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。
    • 7. 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。
    • 8. 发明申请
    • Video Conference Rate Matching
    • 视频会议费率匹配
    • US20100153574A1
    • 2010-06-17
    • US12334969
    • 2008-12-15
    • Ming-Chieh LeeMu HanTim Moore
    • Ming-Chieh LeeMu HanTim Moore
    • G06F15/16
    • H04N7/15H04L12/1827H04N21/25833H04N21/44029H04N21/4788H04N21/6377H04N21/658
    • Video conference rate matching may be provided. A video conference server may receive video source streams from clients on a video conference. The server may analyze each client's capabilities and choose a video stream to send to each client based on those capabilities. For example, a client capable of encoding and decoding a high definition video stream may provide three source video streams—a high definition stream, a medium resolution stream, and a low resolution stream. The server may send only the low resolution stream to a client with a low amount of available bandwidth. The server may send the medium resolution stream to another client with sufficient bandwidth for the high definition stream, but which lacks the ability to decode the high definition stream. choosing at least one of the plurality of data streams to send to the at least one of the plurality of clients based on the analyzed at least one capability; and sending the chosen at least one of the plurality of data streams to the at least one of the plurality of clients.
    • 可以提供视频会议速率匹配。 视频会议服务器可以在视频会议上从客户接收视频源流。 服务器可以分析每个客户端的功能,并根据这些功能选择一个视频流来发送给每个客户端。 例如,能够对高分辨率视频流进行编码和解码的客户端可以提供三个源视频流 - 高清晰度流,中分辨率流和低分辨率流。 服务器只能将低分辨率流发送到具有低可用带​​宽的客户端。 服务器可以向具有足够带宽的高清晰度流的另一个客户端发送媒体分辨率流,但是缺少解码高分辨率流的能力。 基于所分析的至少一个能力来选择所述多个数据流中的至少一个以发送到所述多个客户端中的至少一个; 以及将所选择的所述多个数据流中的至少一个发送到所述多个客户端中的至少一个。
    • 10. 发明申请
    • Infrastructure for enabling high quality real-time audio
    • 实现高质量实时音频的基础设施
    • US20070115949A1
    • 2007-05-24
    • US11281113
    • 2005-11-17
    • Mu HanWarren BarkleyWei ZhongGurdeep Pall
    • Mu HanWarren BarkleyWei ZhongGurdeep Pall
    • H04L12/66
    • H04L29/06027H04L65/103H04L65/104H04L65/605H04L65/80
    • Various technologies and techniques are disclosed that improve media communications. In one embodiment, a media server receives a media communication with a first quality from a personal computer with VoIP telephone capabilities. The media server translates the media (e.g., audio, visual, etc.) communication into a second quality, and forwards the media communication to a communication gateway. The translation to improve communications can also be done when receiving the media communication from the communication gateway for forwarding to the persona computer having VoIP telephone capability. In some embodiments, a media server sits in the communication channel between a personal computer with VoIP telephone capabilities and a communication gateway and is able to translate communications into codec protocols they each understand.
    • 公开了改进媒体通信的各种技术和技术。 在一个实施例中,媒体服务器从具有VoIP电话能力的个人计算机接收具有第一质量的媒体通信。 媒体服务器将媒体(例如,音频,视频等)通信转换成第二质量,并将媒体通信转发到通信网关。 当从通信网关接收媒体通信以转发到具有VoIP电话能力的个人计算机时,也可以进行改善通信的转换。 在一些实施例中,媒体服务器位于具有VoIP电话能力的个人计算机和通信网关之间的通信信道中,并且能够将通信转换为他们都理解的编解码器协议。