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1. 发明授权

US08200681B2 Collaborative media recommendation and sharing technique 有权
标题翻译：合作媒体推荐和分享技术
公开(公告)号：US08200681B2
公开(公告)日：2012-06-12
申请号：US11843129
申请日：2007-08-22
申请人： Rajesh Hedge , Zicheng Liu , Li-wei He , Philip Chou , Christopher Meek
发明人： Rajesh Hedge , Zicheng Liu , Li-wei He , Philip Chou , Christopher Meek
IPC分类号： G06F7/00 , G06F17/30
CPC分类号： G06F17/30029 , G06F17/30053
摘要： A media recommendation and sharing technique that employs agents on media players/devices to expand the scope of media sharing scenarios. The technique assists a user in discovering media items, such as, for example, music, recordings, play lists, pictures, video games, on nearby media players or devices (devices which are capable of receiving, storing and playing media) which are interesting to the user. The collaborative media recommendation and sharing technique contemporaneously determines a user's media preferences based on media stored on a pair of media devices and recommends media for potential sharing based on these determined user preferences.
摘要翻译：媒体推荐和分享技术，在媒体播放器/设备上使用代理扩展媒体共享场景的范围。该技术帮助用户发现媒体项目，例如音乐，录音，播放列表，图片，视频游戏，附近的媒体播放器或设备（能够接收，存储和播放媒体的设备），这是有趣的给用户合作媒体推荐和共享技术同时基于存储在一对媒体设备上的媒体同时确定用户的媒体偏好，并且基于这些确定的用户偏好推荐媒体进行潜在共享。

2. 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。

3. 发明授权

US07337108B2 System and method for providing high-quality stretching and compression of a digital audio signal 有权
标题翻译：用于提供数字音频信号的高质量拉伸和压缩的系统和方法
公开(公告)号：US07337108B2
公开(公告)日：2008-02-26
申请号：US10660325
申请日：2003-09-10
申请人： Dinei Florencio , Philip Chou , Li-Wei He
发明人： Dinei Florencio , Philip Chou , Li-Wei He
IPC分类号： G10L11/06 , G10L21/04 , H04B1/66
CPC分类号： G10L21/04 , G10L2025/935
摘要： An adaptive “temporal audio scaler” is provided for automatically stretching and compressing frames of audio signals received across a packet-based network. Prior to stretching or compressing segments of a current frame, the temporal audio scaler first computes a pitch period for each frame for sizing signal templates used for matching operations in stretching and compressing segments. Further, the temporal audio scaler also determines the type or types of segments comprising each frame. These segment types include “voiced” segments, “unvoiced” segments, and “mixed” segments which include both voiced and unvoiced portions. The stretching or compression methods applied to segments of each frame are then dependent upon the type of segments comprising each frame. Further, the amount of stretching and compression applied to particular segments is automatically variable for minimizing signal artifacts while still ensuring that an overall target stretching or compression ratio is maintained for each frame.
摘要翻译：提供了一种自适应“时间音频缩放器”，用于自动地拉伸和压缩通过基于分组的网络接收的音频信号的帧。在拉伸或压缩当前帧的段之前，时间音频缩放器首先计算用于每个帧的音调周期，用于调整用于拉伸和压缩段中的匹配操作的信号模板。此外，时间音频缩放器还确定包括每个帧的片段的类型或类型。这些段类型包括“有声”段，“无声”段和包括有声和无声部分的“混合”段。然后，应用于每个帧的段的拉伸或压缩方法取决于包括每个帧的段的类型。此外，施加到特定段的拉伸和压缩量可自动变化以最小化信号伪影，同时仍然确保为每个帧维持整体目标拉伸或压缩比。

4. 发明申请

US20050058145A1 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：US20050058145A1
公开(公告)日：2005-03-17
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： G10L19/00 , H04L12/56
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。

5. 发明申请

US20050055204A1 System and method for providing high-quality stretching and compression of a digital audio signal 有权
标题翻译：用于提供数字音频信号的高质量拉伸和压缩的系统和方法
公开(公告)号：US20050055204A1
公开(公告)日：2005-03-10
申请号：US10660325
申请日：2003-09-10
申请人： Dinei Florencio , Philip Chou , Li-Wei He
发明人： Dinei Florencio , Philip Chou , Li-Wei He
IPC分类号： G10L19/00 , G01L19/00 , G10L11/06 , G10L21/04 , H03M7/30 , G10L15/00
CPC分类号： G10L21/04 , G10L2025/935
摘要： An adaptive “temporal audio scaler” is provided for automatically stretching and compressing frames of audio signals received across a packet-based network. Prior to stretching or compressing segments of a current frame, the temporal audio scaler first computes a pitch period for each frame for sizing signal templates used for matching operations in stretching and compressing segments. Further, the temporal audio scaler also determines the type or types of segments comprising each frame. These segment types include “voiced” segments, “unvoiced” segments, and “mixed” segments which include both voiced and unvoiced portions. The stretching or compression methods applied to segments of each frame are then dependent upon the type of segments comprising each frame. Further, the amount of stretching and compression applied to particular segments is automatically variable for minimizing signal artifacts while still ensuring that an overall target stretching or compression ratio is maintained for each frame.
摘要翻译：提供了一种自适应“时间音频缩放器”，用于自动地拉伸和压缩通过基于分组的网络接收的音频信号的帧。在拉伸或压缩当前帧的段之前，时间音频缩放器首先计算用于每个帧的音调周期，用于调整用于拉伸和压缩段中的匹配操作的信号模板。此外，时间音频缩放器还确定包括每个帧的片段的类型或类型。这些段类型包括“有声”段，“无声”段和包括有声和无声部分的“混合”段。然后，应用于每个帧的段的拉伸或压缩方法取决于包括每个帧的段的类型。此外，施加到特定段的拉伸和压缩量可自动变化以最小化信号伪影，同时仍然确保为每个帧维持整体目标拉伸或压缩比。

6. 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。

7. 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。

8. 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。

9. 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。

10. 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。

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