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1. 发明授权

US07697514B2 Receiver-driven layered error correction multicast over heterogeneous packet networks 有权
标题翻译：接收器驱动的分层纠错多播在异构分组网络上
公开(公告)号：US07697514B2
公开(公告)日：2010-04-13
申请号：US11109250
申请日：2005-04-18
申请人： Philip Chou , Albert Wang , Sanjeev Mehrotra , Alexander Mohr
发明人： Philip Chou , Albert Wang , Sanjeev Mehrotra , Alexander Mohr
IPC分类号： H04L12/56
CPC分类号： H04L1/007 , H04L1/0009 , H04L1/0017 , H04L1/06 , H04L1/1812 , H04L2001/0093
摘要： 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.
摘要翻译：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

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。

3. 发明申请

US20070237130A1 Providing contextual information with a voicemail message 有权
标题翻译：使用语音邮件消息提供上下文信息
公开(公告)号：US20070237130A1
公开(公告)日：2007-10-11
申请号：US11398816
申请日：2006-04-06
申请人： David Milstein , David Howell , Linda Criddle , Philip Chou , Timothy Moore
发明人： David Milstein , David Howell , Linda Criddle , Philip Chou , Timothy Moore
IPC分类号： H04L12/66
CPC分类号： H04M3/53366 , H04M2203/303
摘要： Aspects of the present invention are directed at obtaining contextual information with a voicemail message. In accordance with one embodiment, a method is provided that obtains additional contextual information that is not obtained automatically when a voicemail message is received. More specifically, the method includes automatically obtaining a first set of contextual information from a client associated with the caller when the caller is transferred to a voicemail system. Then a determination is made regarding whether the callee requests that additional contextual information be obtained. If the callee requests that additional contextual information be obtained, the requested information is obtained from a third-party service or by prompting the caller.
摘要翻译：本发明的方面涉及通过语音邮件消息获取上下文信息。根据一个实施例，提供了一种获得当接收到语音邮件消息时不自动获得的附加语境信息的方法。更具体地，该方法包括当主叫方被转移到语音邮件系统时，从与呼叫者相关联的客户端自动获得第一组上下文信息。然后确定被叫方是否请求获得附加的上下文信息。如果被叫方请求获得附加的上下文信息，则从第三方服务或通过提示呼叫者获得所请求的信息。

4. 发明申请

US20060251062A1 Scalable overlay network 有权
公开(公告)号：US20060251062A1
公开(公告)日：2006-11-09
申请号：US11101121
申请日：2005-04-07
申请人： Kamal Jain , Laszlo Lovasz , Philip Chou
发明人： Kamal Jain , Laszlo Lovasz , Philip Chou
IPC分类号： H04L12/56
CPC分类号： H04L45/02 , H04L12/2856 , H04L67/16
摘要： An overlay network and scheme for building and using the overlay network are described. As the overlay network is built, new nodes joining the network are connected randomly with other nodes which results in a random graph as the network topology. The network is fully scalable, with each new node providing the same number of network connections for output to other nodes as it consumes when it joins the network. In addition, network coding is implemented at each node to mix packets of data entering each node using random linear functions. The network coding at each node generates new packets that are independent combinations of original data packets entering the node. The new coded packets make the distribution of data to other nodes more efficient and robust.

5. 发明申请

US20060198446A1 Generalized reference decoder for image or video processing 有权
公开(公告)号：US20060198446A1
公开(公告)日：2006-09-07
申请号：US11418995
申请日：2006-05-04
申请人： Jordi Ribas-Corbera , Philip Chou
发明人： Jordi Ribas-Corbera , Philip Chou
IPC分类号： H04N11/02 , H04N11/04 , H04N7/12 , H04B1/66
CPC分类号： H04N19/44 , H04N19/115 , H04N19/149 , H04N19/152 , H04N19/172 , H04N19/61
摘要： A method and system including an improved generalized reference decoder that operates according to any number of sets of rate and buffer parameters for a given bit stream. Each set characterizes a leaky bucket model and contains three parameters representing the transmission bit rate, buffer size, and initial decoder buffer fullness. An encoder provides at least two sets of these parameters, whereby the decoder selects one or interpolates between them to operate at any desired peak bit rate, buffer size or delay. The generalized reference decoder may select the smallest buffer size and corresponding delay that decodes the bit stream without buffer underflow or overflow, or alternatively may select and operate at the minimum required peak transmission rate, or something between both. In practice, the buffer size, delay and/or the peak transmission rate can be reduced by significant factors, and/or the signal-to-noise ratio (SNR) can be increased.

6. 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.

7. 发明申请

US20050055201A1 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：US20050055201A1
公开(公告)日：2005-03-10
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： G10L11/02 , G10L11/06
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。

8. 发明授权

US5602589A Video image compression using weighted wavelet hierarchical vector quantization 失效
标题翻译：使用加权小波分层矢量量化的视频图像压缩
公开(公告)号：US5602589A
公开(公告)日：1997-02-11
申请号：US293612
申请日：1994-08-19
申请人： Mohan Vishwanath , Philip Chou
发明人： Mohan Vishwanath , Philip Chou
IPC分类号： H04N7/30 , G06T9/00 , H03M7/40 , H04N7/26 , H04N7/28
CPC分类号： G06T9/008 , H04N19/40 , H04N19/62 , H04N19/63 , H04N19/90 , H04N19/94 , H04N19/42
摘要： A weighted wavelet hierarchical vector quantization (WWHVQ) procedure is initiated by obtaining an N.times.N pixel image where 8 bits per pixel (steps 10 and 12). A look-up operation is performed to obtain data representing a discrete wavelet transform (DWT) followed by a quantization of the data (step 14). Upon completion of the look-up, a data compression will have been performed. Further stages and look-up will result in further compression of the data, i.e., 4:1, 8:1, 16:1, 32:1, 64:1, . . . etc. Accordingly, a determination is made whether the compression is complete (step 16). If the compression is incomplete, further look-up is performed. If the compression is complete, however, the compressed data is transmitted (step 18). It is determined at a gateway whether further compression is required (step 19). If so, transcoding is performed (step 20). The receiver receives the compressed data (step 22). Subsequently, a second look-up operation is performed to obtain data representing an inverse discrete wavelet transform of the decompressed data (step 24). After one iteration, the data is decompressed by a factor of two. Further iterations allows for further decompression of the data. Accordingly, a determination is made whether decompression is complete (step 26). If the decompression is in incomplete, further look-ups are performed. If, however, the decompression is complete, the WWHVQ procedure is ended (step 28).
摘要翻译：通过获得每像素8位（步骤10和12）的N×N像素图像来启动加权小波分层矢量量化（WWHVQ）过程。执行查找操作以获得表示离散小波变换（DWT）的数据，随后数据的量化（步骤14）。完成查找后，将执行数据压缩。进一步的阶段和查找将导致数据的进一步压缩，即4：1,8：1,16：1，32：1，64：1。。。因此，确定压缩是否完成（步骤16）。如果压缩不完整，则进一步查找。然而，如果压缩完成，则传送压缩数据（步骤18）。在网关确定是否需要进一步的压缩（步骤19）。如果是，则执行代码转换（步骤20）。接收器接收压缩数据（步骤22）。随后，执行第二查找操作以获得表示解压缩数据的逆离散小波变换的数据（步骤24）。一次迭代后，数据被解压缩一倍。进一步的迭代允许进一步解压缩数据。因此，确定减压是否完成（步骤26）。如果解压缩不完整，则进行进一步的查找。然而，如果解压缩完成，则WWHVQ过程结束（步骤28）。

9. 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。

10. 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。

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