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1. 发明授权

US07543073B2 System and process for performing an exponentially weighted moving average on streaming data to establish a moving average bit rate 有权
标题翻译：用于在流数据上执行指数加权移动平均以建立移动平均比特率的系统和过程
公开(公告)号：US07543073B2
公开(公告)日：2009-06-02
申请号：US11010040
申请日：2004-12-10
申请人： Philip Chou , Anders Klemets , Cheng Huang
发明人： Philip Chou , Anders Klemets , Cheng Huang
IPC分类号： G06F15/16
CPC分类号： H04N21/44004 , H04L47/10 , H04N21/23406 , H04N21/2662 , H04N21/6373 , H04N21/6379 , H04N21/6547
摘要： A system and process for performing an exponentially weighted moving average on streaming data to establish a moving average bit rate of data units is presented. In general, the system or process computes, on a unit-by-unit basis, the product of the moving average bit rate computed for a data unit immediately prior to a unit under consideration and a first fractional weighting factor, added to the product of the instantaneous bit rate of the data unit under consideration and a second fractional weighting factor, wherein at least one fractional weighting factor is not a constant but instead based on the time between data units.
摘要翻译：提出了一种用于在流数据上执行指数加权移动平均以建立数据单元的移动平均比特率的系统和过程。一般来说，系统或过程在逐单位的基础上计算紧接在所考虑的单元之前针对数据单元计算的移动平均比特率和第一分数加权因子的乘积，所考虑的数据单元的瞬时比特率和第二分数加权因子，其中至少一个分数加权因子不是常数，而是基于数据单元之间的时间。

2. 发明申请

US20060126713A1 System and process for performing an exponentially weighted moving average on streaming data to establish a moving average bit rate 有权
公开(公告)号：US20060126713A1
公开(公告)日：2006-06-15
申请号：US11010040
申请日：2004-12-10
申请人： Philip Chou , Anders Klemets , Cheng Huang
发明人： Philip Chou , Anders Klemets , Cheng Huang
IPC分类号： H04B17/00
CPC分类号： H04N21/44004 , H04L47/10 , H04N21/23406 , H04N21/2662 , H04N21/6373 , H04N21/6379 , H04N21/6547
摘要： A system and process for performing an exponentially weighted moving average on streaming data to establish a moving average bit rate of data units is presented. In general, the system or process computes, on a unit-by-unit basis, the product of the moving average bit rate computed for a data unit immediately prior to a unit under consideration and a first fractional weighting factor, added to the product of the instantaneous bit rate of the data unit under consideration and a second fractional weighting factor, wherein at least one fractional weighting factor is not a constant but instead based on the time between data units.

3. 发明授权

US07536469B2 System and process for controlling the coding bit rate of streaming media data employing a limited number of supported coding bit rates 有权
标题翻译：使用有限数量的支持的编码比特率来控制流媒体数据的编码比特率的系统和过程
公开(公告)号：US07536469B2
公开(公告)日：2009-05-19
申请号：US11010113
申请日：2004-12-10
申请人： Philip Chou , Anders Klemets , Cheng Huang
发明人： Philip Chou , Anders Klemets , Cheng Huang
IPC分类号： G06F15/16
CPC分类号： H04N21/44004 , H04N21/23406 , H04N21/2402 , H04N21/44245 , H04N21/6373 , H04N21/643
摘要： A system and process for controlling the coding bit rate of streaming media data is presented where a server streams data that exhibits one of a number of coding bit rates supported by the server. Initially, the server chooses the coding bit rate. However, after this startup period, the client provides coding bit rate requests. The server transmits the streaming media data at the most appropriate supported coding bit rate closest to the rate requested. The coding bit rates requested are those estimated to provide a high quality playback of the streaming data while still keeping a decoder buffer of the client filled to a desired level. A leaky bucket model is incorporated so that the changes in buffer duration due to natural variation in the instantaneous coding bit rate are not mistaken for changes in buffer duration due to network congestion.
摘要翻译：呈现用于控制流媒体数据的编码比特率的系统和过程，其中服务器流传送由服务器支持的多个编码比特率之一的数据。最初，服务器选择编码比特率。然而，在这个启动期之后，客户端提供编码比特率请求。服务器以最接近请求速率的最适合的支持的编码比特率来发送流媒体数据。所要求的编码比特率是估计提供流式数据的高质量回放同时仍然保持客户端的解码器缓冲器被填充到期望水平的编码比特率。引入泄漏桶模型，使得由于瞬时编码比特率的自然变化而导致的缓冲持续时间的变化不会由于网络拥塞而被误认为缓冲持续时间的变化。

4. 发明申请

US20060165166A1 System and process for controlling the coding bit rate of streaming media data employing a limited number of supported coding bit rates 有权
公开(公告)号：US20060165166A1
公开(公告)日：2006-07-27
申请号：US11010113
申请日：2004-12-10
申请人： Philip Chou , Anders Klemets , Cheng Huang
发明人： Philip Chou , Anders Klemets , Cheng Huang
IPC分类号： H04N7/12 , H04B1/66
CPC分类号： H04N21/44004 , H04N21/23406 , H04N21/2402 , H04N21/44245 , H04N21/6373 , H04N21/643
摘要： A system and process for controlling the coding bit rate of streaming media data is presented where a server streams data that exhibits one of a number of coding bit rates supported by the server. Initially, the server chooses the coding bit rate. However, after this startup period, the client provides coding bit rate requests. The server transmits the streaming media data at the most appropriate supported coding bit rate closest to the rate requested. The coding bit rates requested are those estimated to provide a high quality playback of the streaming data while still keeping a decoder buffer of the client filled to a desired level. A leaky bucket model is incorporated so that the changes in buffer duration due to natural variation in the instantaneous coding bit rate are not mistaken for changes in buffer duration due to network congestion.

5. 发明申请

US20060143678A1 System and process for controlling the coding bit rate of streaming media data employing a linear quadratic control technique and leaky bucket model 审中-公开
标题翻译：使用线性二次控制技术和漏斗模型来控制流媒体数据的编码比特率的系统和过程
公开(公告)号：US20060143678A1
公开(公告)日：2006-06-29
申请号：US11010114
申请日：2004-12-10
申请人： Philip Chou , Anders Klemets , Cheng Huang
发明人： Philip Chou , Anders Klemets , Cheng Huang
IPC分类号： H04N7/173
CPC分类号： H04N21/44004 , H04N19/115 , H04N19/149 , H04N19/152 , H04N19/46 , H04N19/61 , H04N21/23406 , H04N21/2343
摘要： A system and process for controlling the coding bit rate of streaming media data is presented. This coding bit rate control involves dynamically adjusting the coding bit rate to control client buffer duration to prevent the buffer from underflowing, while keeping the average coding bit rate close to the average transmission bit rate of the network (an thus maximizing the quality of the data playback). Using the theory of optimal linear quadratic control, the client buffer duration is kept as close as possible to a target level while still keeping the coding bit rate (and hence the quality) as constant as possible. In addition, a leaky bucket model is incorporated into the control loop so that the changes in buffer duration due to natural variation in the instantaneous coding bit rate are not mistaken for changes in buffer duration due to network congestion.
摘要翻译：提出了一种用于控制流媒体数据的编码比特率的系统和过程。该编码比特率控制涉及动态调整编码比特率以控制客户端缓冲持续时间以防止缓冲器下溢，同时保持平均编码比特率接近于网络的平均传输比特率（从而使数据质量最大化回放）。使用最优线性二次控制理论，尽可能保持编码比特率（因此质量）尽可能保持尽可能接近目标级别的客户端缓冲持续时间。另外，泄漏桶模型被并入到控制回路中，使得由于瞬时编码比特率的自然变化而导致的缓冲持续时间的变化不会被误认为由于网络拥塞而导致的缓冲持续时间的变化。

6. 发明申请

US20070206610A1 Scheduling Connections Between Peers in a Peer-to-Peer File Sharing Environment 有权
标题翻译：在对等文件共享环境中调度对等体之间的连接
公开(公告)号：US20070206610A1
公开(公告)日：2007-09-06
申请号：US11275177
申请日：2005-12-16
申请人： Dan Teodosiu , Philip Chou , Alexandra Heron , Cheng Huang , Tom Kleinpeter , Jin Li , Eyal Schwartz , John Spivey , David Steere
发明人： Dan Teodosiu , Philip Chou , Alexandra Heron , Cheng Huang , Tom Kleinpeter , Jin Li , Eyal Schwartz , John Spivey , David Steere
IPC分类号： H04L12/56 , H04J3/06
CPC分类号： H04L67/325 , H04L67/06 , H04L67/104 , H04L67/1044 , H04L67/1095 , H04W76/14
摘要： Various embodiments can utilize a distributed solution for scheduling connections between peers in a file sharing environment. In accordance with at least some embodiments, individual nodes in a peer-to-peer system include scheduling software that enables the nodes to make scheduling decisions with regard to how, when and where connections are made with other peers. Scheduling decisions can be based on a number of different parameters. In at least some embodiments, a synchronization negotiation manager is embodied with logic that drives the negotiation and scheduling process. This logic is represented by an innovative state machine that is designed to implement the negotiation and synchronization process.
摘要翻译：各种实施例可以利用分布式解决方案来调度文件共享环境中的对等体之间的连接。根据至少一些实施例，对等系统中的单个节点包括调度软件，其使节点能够就如何，何时何地与其他对等体进行连接做出调度决定。调度决定可以基于许多不同的参数。在至少一些实施例中，同步协商管理器具有驱动协商和调度过程的逻辑。该逻辑由创新的状态机表示，旨在实现协商和同步过程。

7. 发明授权

US07596488B2 System and method for real-time jitter control and packet-loss concealment in an audio signal 有权
标题翻译：用于音频信号中实时抖动控制和丢包隐藏的系统和方法
公开(公告)号：US07596488B2
公开(公告)日：2009-09-29
申请号：US10663390
申请日：2003-09-15
申请人： Dinei Florencio , Philip Chou , Li-Wei He
发明人： Dinei Florencio , Philip Chou , Li-Wei He
IPC分类号： G10L19/04 , G10L21/04 , G10L11/06
CPC分类号： H04N21/4392 , G10L19/005 , H04N21/4394 , H04N21/4396 , H04N21/4398
摘要： An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.
摘要翻译： “自适应音频播放控制器”通过将音频信号的接收分组解码并读取到信号缓冲器来进行操作。然后根据播放器设备的需要从信号缓冲器中播放经解码的音频信号的样本。抖动控制和分组丢失隐藏是通过实时连续分析缓冲区内容来实现的，并且确定是否从缓冲器内容中提供未修改的重放，是否压缩缓冲区内容，扩展缓冲区内容，还是提供丢包隐藏延迟或丢失的数据包作为缓冲区内容的函数。此外，自适应音频重放控制器还确定在哪里拉伸或压缩信号缓冲器中的特定帧或信号段，以及拉伸或压缩这些段以便优化感知的播放质量。

8. 发明申请

US20050185578A1 Methods and systems for streaming data 有权
标题翻译：流数据的方法和系统
公开(公告)号：US20050185578A1
公开(公告)日：2005-08-25
申请号：US10787612
申请日：2004-02-25
申请人： Venkata Padmanabhan , Jiahe Wang , Philip Chou
发明人： Venkata Padmanabhan , Jiahe Wang , Philip Chou
IPC分类号： G01R31/08 , H04L12/18 , H04L12/56
CPC分类号： H04L12/185 , H04L41/5009 , H04L41/5035 , H04L43/0829 , H04L47/10 , H04L47/11 , H04L47/12 , H04L67/104 , H04L67/1089
摘要： A technique is disclosed that can efficiently control congestion, while supporting heterogeneity for streaming data among multiple computers in a network. A plurality of nodes is divided into a plurality of distribution trees within a computer network, wherein the data is divided into a plurality of prioritized layers. When a node experiences packet loss, the location of the congestion is inferred. If the congestion is at or near the outgoing link, outgoing traffic is shed to alleviate the congestion by shedding child node(s) receiving descriptions in the least important layer of data that the child node(s) are receiving. Similarly, if the congestion is at or near the incoming link, incoming traffic is shed by shedding parent nodes that are sending descriptions in the least important layer of data that the node is receiving. Nodes with available bandwidth are further instructed to subscribe to additional descriptions.
摘要翻译：公开了一种可以有效地控制拥塞的技术，同时支持网络中的多个计算机之间的流数据的异构性。多个节点被划分成计算机网络内的多个分配树，其中数据被分成多个优先化层。当节点遇到数据包丢失时，推断出拥塞的位置。如果拥塞处于或接近输出链路，则流出流量被减轻，以减轻子节点在子节点正在接收的最不重要的数据层中接收描述的缓冲来减轻拥塞。类似地，如果拥塞处于或接近传入链路，则通过在发送节点正在接收的最不重要的数据层中发送描述的父节点脱离传入流量。进一步指示具有可用带宽的节点订阅附加描述。

9. 发明授权

US07412376B2 System and method for real-time detection and preservation of speech onset in a signal 有权
标题翻译：用于实时检测和保存信号中语音发生的系统和方法
公开(公告)号：US07412376B2
公开(公告)日：2008-08-12
申请号：US10660326
申请日：2003-09-10
申请人： Dinei Florencio , Philip Chou
发明人： Dinei Florencio , Philip Chou
IPC分类号： G10L11/00 , G10L11/02
CPC分类号： G10L25/87 , G10L2025/783
摘要： A “speech onset detector” provides a variable length frame buffer in combination with either variable transmission rate or temporal speech compression for buffered signal frames. The variable length buffer buffers frames that are not clearly identified as either speech or non-speech frames during an initial analysis. Buffering of signal frames continues until a current frame is identified as either speech or non-speech. If the current frame is identified as non-speech, buffered frames are encoded as non-speech frames. However, if the current frame is identified as a speech frame, buffered frames are searched for the actual onset point of the speech. Once that onset point is identified, the signal is either transmitted in a burst, or a time-scale modification of the buffered signal is applied for compressing buffered frames beginning with the frame in which onset point is detected. The compressed frames are then encoded as one or more speech frames.
摘要翻译： “语音起始检测器”提供了可变长度帧缓冲器，与缓冲信号帧的可变传输速率或时间语音压缩相结合。可变长度缓冲器缓冲在初始分析期间未被清楚地识别为语音或非语音帧的帧。信号帧的缓冲持续到当前帧被识别为语音或非语音。如果当前帧被识别为非语音，则缓冲帧被编码为非语音帧。然而，如果当前帧被识别为语音帧，则搜索缓冲的帧用于语音的实际起始点。一旦该起始点被识别，则信号以突发方式发送，或者缓冲信号的时间尺度修改被应用于从检测到起始点的帧开始的缓冲帧。然后将压缩的帧编码为一个或多个语音帧。

10. 发明申请

US20070263607A1 Extensible metadata structure 失效
标题翻译：可扩展元数据结构
公开(公告)号：US20070263607A1
公开(公告)日：2007-11-15
申请号：US11394773
申请日：2006-03-31
申请人： David Milstein , David Howell , Linda Criddle , Michael Malueg , Philip Chou
发明人： David Milstein , David Howell , Linda Criddle , Michael Malueg , Philip Chou
IPC分类号： H04L12/66
CPC分类号： H04L12/66
摘要： Structured hierarchies for communicating contextual information relating to a VoIP conversation are provided. The structured hierarchies are utilized for efficient communications of various amounts and types of contextual information over a VoIP conversation channel. Information identifying at least one structured hierarchy, which will be used to carry the contextual information, is transmitted during establishment of a conversation between two VoIP enhanced devices and prior to the exchange of contextual information. The structural hierarchy is selected from a set of predefined and declared structured hierarchies. Subsequently transmitted contextual information exchanged between two VoIP enhanced devices is represented in accordance with the identified structural hierarchy. Additionally, the structural hierarchies can be extensible by the addition of more definitions to the current structural hierarchies.
摘要翻译：提供了用于传送与VoIP会话相关的上下文信息的结构化层级。结构化层次被用于通过VoIP对话信道有效地通信各种数量和类型的上下文信息。在两个VoIP增强设备之间的交谈建立之前，以及交换上下文信息之前，发送识别用于携带上下文信息的至少一个结构化层次结构的信息。从一组预定义和声明的结构化层次结构中选择结构层次结构。随后根据所识别的结构层次来表示在两个VoIP增强设备之间交换的传送的上下文信息。此外，通过向当前的结构层次结构添加更多的定义，结构层次结构可以扩展。

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