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    • 1. 发明授权
    • Receiver-driven layered error correction multicast over heterogeneous packet networks
    • 接收器驱动的分层纠错多播在异构分组网络上
    • US07697514B2
    • 2010-04-13
    • US11109250
    • 2005-04-18
    • Philip ChouAlbert WangSanjeev MehrotraAlexander Mohr
    • Philip ChouAlbert WangSanjeev MehrotraAlexander Mohr
    • H04L12/56
    • H04L1/007H04L1/0009H04L1/0017H04L1/06H04L1/1812H04L2001/0093
    • A system and method for correcting errors and losses occurring during a receiver-driven layered multicast (RLM) of real-time media over a heterogeneous packet network such as the Internet. This is accomplished by augmenting RLM with one or more layers of error correction information. This allows each receiver to separately optimize the quality of received audio and video information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM would have one or more multicasted error correction data streams (i.e., layers) associated therewith. Each of the error correction layers would contain information that can be used to replace lost packets from the associated source layer. More than one error correction layer is proposed as some of the error correction packets contained in the data stream needed to replace the packets lost in the associated source stream may themselves be lost in transmission. A preferred process for generating the error correction streams involves the use of a unique adaptation of the Forward Error Correction (FEC) techniques. This process encodes the transmission data using a linear transform which adds redundant elements. The redundancy permits losses to be corrected because any of the original data elements can be derived from any of the encoded elements. Thus, as long as enough of the encoded data elements are received so as to equal the number of the original data elements, it is possible to derive all the original elements.
    • 一种用于在异构分组网络(例如因特网)下校正在实时媒体的接收机驱动分层多播(RLM)期间发生的错误和损失的系统和方法。 这是通过用一层或多层纠错信息增强RLM来实现的。 这允许每个接收机通过订阅至少一个纠错层来分别优化所接收的音频和视频信息的质量。 理想地,RLM中的每个源层将具有与其相关联的一个或多个多播的纠错数据流(即,层)。 每个纠错层将包含可用于替换相关源层丢失的分组的信息。 提出了多于一个纠错层,因为包含在替换相关源流中丢失的分组所需的数据流中的一些纠错分组本身可能在传输中丢失。 用于产生纠错流的优选过程涉及使用前向纠错(FEC)技术的唯一适配。 该过程使用添加冗余元素的线性变换对传输数据进行编码。 冗余允许修正损失,因为任何原始数据元素可以从任何编码元素导出。 因此,只要接收到足够的编码数据元素以便等于原始数据元素的数量,就有可能导出所有的原始元素。
    • 2. 发明授权
    • Receiver-driven layered error correction multicast over heterogeneous packet networks
    • 接收器驱动的分层纠错多播在异构分组网络上
    • US07366172B2
    • 2008-04-29
    • US11177258
    • 2005-07-08
    • Philip ChouAlbert WangSanjeev MehrotraAlexander Mohr
    • Philip ChouAlbert WangSanjeev MehrotraAlexander Mohr
    • H04L12/56
    • H04L1/007H04L1/0009H04L1/0017H04L1/06H04L1/1812H04L2001/0093
    • A system and method for correcting errors and losses occurring during a receiver-driven layered multicast (RLM) of real-time media over a heterogeneous packet network such as the Internet. This is accomplished by augmenting RLM with one or more layers of error correction information. This allows each receiver to separately optimize the quality of received audio and video information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM would have one or more multicasted error correction data streams (i.e., layers) associated therewith. Each of the error correction layers would contain information that can be used to replace lost packets from the associated source layer. More than one error correction layer is proposed as some of the error correction packets contained in the data stream needed to replace the packets lost in the associated source stream may themselves be lost in transmission. A preferred process for generating the error correction streams involves the use of a unique adaptation of the Forward Error Correction (FEC) techniques. This process encodes the transmission data using a linear transform which adds redundant elements. The redundancy permits losses to be corrected because any of the original data elements can be derived from any of the encoded elements. Thus, as long as enough of the encoded data elements are received so as to equal the number of the original data elements, it is possible to derive all the original elements.
    • 一种用于在异构分组网络(例如因特网)下校正在实时媒体的接收机驱动分层多播(RLM)期间发生的错误和损失的系统和方法。 这是通过用一层或多层纠错信息增强RLM来实现的。 这允许每个接收机通过订阅至少一个纠错层来分别优化所接收的音频和视频信息的质量。 理想地,RLM中的每个源层将具有与其相关联的一个或多个多播的纠错数据流(即,层)。 每个纠错层将包含可用于替换相关源层丢失的分组的信息。 提出了多于一个纠错层,因为包含在替换相关源流中丢失的分组所需的数据流中的一些纠错分组本身可能在传输中丢失。 用于产生纠错流的优选过程涉及使用前向纠错(FEC)技术的唯一适配。 该过程使用添加冗余元素的线性变换对传输数据进行编码。 冗余允许修正损失,因为任何原始数据元素可以从任何编码元素导出。 因此,只要接收到足够的编码数据元素以便等于原始数据元素的数量,就有可能导出所有的原始元素。
    • 4. 发明申请
    • Receiver-driven layered error correction multicast over heterogeneous packet networks
    • 接收器驱动的分层纠错多播在异构分组网络上
    • US20050249211A1
    • 2005-11-10
    • US11177258
    • 2005-07-08
    • Philip ChouAlbert WangSanjeev MehrotraAlexander Mohr
    • Philip ChouAlbert WangSanjeev MehrotraAlexander Mohr
    • H04L1/00H04L1/06H04L1/08G06K5/04H04J3/24H04L12/28H04L12/56
    • H04L1/007H04L1/0009H04L1/0017H04L1/06H04L1/1812H04L2001/0093
    • A system and method for correcting errors and losses occurring during a receiver-driven layered multicast (RLM) of real-time media over a heterogeneous packet network such as the Internet. This is accomplished by augmenting RLM with one or more layers of error correction information. This allows each receiver to separately optimize the quality of received audio and video information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM would have one or more multicasted error correction data streams (i.e., layers) associated therewith. Each of the error correction layers would contain information that can be used to replace lost packets from the associated source layer. More than one error correction layer is proposed as some of the error correction packets contained in the data stream needed to replace the packets lost in the associated source stream may themselves be lost in transmission. A preferred process for generating the error correction streams involves the use of a unique adaptation of the Forward Error Correction (FEC) techniques. This process encodes the transmission data using a linear transform which adds redundant elements. The redundancy permits losses to be corrected because any of the original data elements can be derived from any of the encoded elements. Thus, as long as enough of the encoded data elements are received so as to equal the number of the original data elements, it is possible to derive all the original elements.
    • 一种用于在异构分组网络(例如因特网)下校正在实时媒体的接收机驱动分层多播(RLM)期间发生的错误和损失的系统和方法。 这是通过用一层或多层纠错信息增强RLM来实现的。 这允许每个接收机通过订阅至少一个纠错层来分别优化所接收的音频和视频信息的质量。 理想地,RLM中的每个源层将具有与其相关联的一个或多个多播的纠错数据流(即,层)。 每个纠错层将包含可用于替换相关源层丢失的分组的信息。 提出了多于一个纠错层,因为包含在替换相关源流中丢失的分组所需的数据流中的一些纠错分组本身可能在传输中丢失。 用于产生纠错流的优选过程涉及使用前向纠错(FEC)技术的唯一适配。 该过程使用添加冗余元素的线性变换对传输数据进行编码。 冗余允许修正损失,因为任何原始数据元素可以从任何编码元素导出。 因此,只要接收到足够的编码数据元素以便等于原始数据元素的数量,就有可能导出所有的原始元素。
    • 5. 发明授权
    • System and method for real-time jitter control and packet-loss concealment in an audio signal
    • 用于音频信号中实时抖动控制和丢包隐藏的系统和方法
    • US07596488B2
    • 2009-09-29
    • US10663390
    • 2003-09-15
    • Dinei FlorencioPhilip ChouLi-Wei He
    • Dinei FlorencioPhilip ChouLi-Wei He
    • G10L19/04G10L21/04G10L11/06
    • H04N21/4392G10L19/005H04N21/4394H04N21/4396H04N21/4398
    • 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.
    • “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。
    • 6. 发明申请
    • Methods and systems for streaming data
    • 流数据的方法和系统
    • US20050185578A1
    • 2005-08-25
    • US10787612
    • 2004-02-25
    • Venkata PadmanabhanJiahe WangPhilip Chou
    • Venkata PadmanabhanJiahe WangPhilip Chou
    • G01R31/08H04L12/18H04L12/56
    • H04L12/185H04L41/5009H04L41/5035H04L43/0829H04L47/10H04L47/11H04L47/12H04L67/104H04L67/1089
    • 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.
    • 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。
    • 8. 发明授权
    • System and method for real-time detection and preservation of speech onset in a signal
    • 用于实时检测和保存信号中语音发生的系统和方法
    • US07412376B2
    • 2008-08-12
    • US10660326
    • 2003-09-10
    • Dinei FlorencioPhilip Chou
    • Dinei FlorencioPhilip Chou
    • G10L11/00G10L11/02
    • G10L25/87G10L2025/783
    • 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.
    • “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
    • Extensible metadata structure
    • 可扩展元数据结构
    • US20070263607A1
    • 2007-11-15
    • US11394773
    • 2006-03-31
    • David MilsteinDavid HowellLinda CriddleMichael MaluegPhilip Chou
    • David MilsteinDavid HowellLinda CriddleMichael MaluegPhilip Chou
    • H04L12/66
    • H04L12/66
    • 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.
    • 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。