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公开(公告)号：US08064470B2
公开(公告)日：2011-11-22
申请号：US10593587
申请日：2005-03-16
IPC分类号： H04L12/28
CPC分类号： H04L65/4084 , H04L47/10 , H04L47/22 , H04L65/607 , H04N19/188 , H04N19/44 , H04N19/61 , H04N21/23406 , H04N21/2401 , H04N21/6332 , H04N21/64322 , H04N21/6587
摘要： Recorded material such as video is transmitted in compressed form to a receiver, which has a buffer for smoothing differences between the data rate received and that consumed by a decoder that follows. The whole of the recording is analysed to determine a point at which to commence playing such that no buffer underflow can occur; the decoder commences playing only when this point has been reached.
摘要翻译： 诸如视频的记录材料以压缩的形式传送到接收机，接收机具有用于平滑接收到的数据速率与随后的解码器消耗的数据速率之间的差异的缓冲器。 分析整个记录以确定开始播放的点，使得不会发生缓冲器下溢; 只有达到这一点，解码器才开始播放。
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公开(公告)号：US20080025340A1
公开(公告)日：2008-01-31
申请号：US10593587
申请日：2005-03-16
IPC分类号： H04L12/413
CPC分类号： H04L65/4084 , H04L47/10 , H04L47/22 , H04L65/607 , H04N19/188 , H04N19/44 , H04N19/61 , H04N21/23406 , H04N21/2401 , H04N21/6332 , H04N21/64322 , H04N21/6587
摘要： Recorded material such as video is transmitted in compressed form to a receiver, which has a buffer for smoothing differences between the data rate received and that consumed by a decoder that follows. The whole of the recording is analysed to determine a point at which to commence playing such that no buffer underflow can occur; the decoder commences playing only when this point has been reached.
摘要翻译： 诸如视频的记录材料以压缩的形式传送到接收机，接收机具有用于平滑接收到的数据速率与随后的解码器消耗的数据速率之间的差异的缓冲器。 分析整个记录以确定开始播放的点，使得不会发生缓冲器下溢; 只有达到这一点，解码器才开始播放。
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公开(公告)号：US07912974B2
公开(公告)日：2011-03-22
申请号：US10549912
申请日：2004-03-23
IPC分类号： G06F15/16
CPC分类号： H04L65/602 , H04L29/06027 , H04L65/80
摘要： Data for presentation in real time, such as a video or audio sequence, is available on different encoded versions having different degrees Of compression. In order to assess, during transmission of one version, the feasibility of switching to another version, given the data rate known to be available at the time, a server computes, for a candidate version, in respect of at least one portion thereof that has not yet been sent, the maximum value of a timing error that would occur if any number of portions starting with that portion to be sent at the available rate. The selection of the same or a different version for continuing transmission is taken in dependence on a comparison between the computed error and the current state of a receiving buffer. Error values may be computed in advance for a range of transmitting rates, stored and later retrieved for use in estimating an error value corresponding to the actual transmitting rate.
摘要翻译： 用于实时呈现的数据，例如视频或音频序列，可在具有不同压缩程度的不同编码版本上使用。 为了在一个版本的传输期间评估切换到另一版本的可行性，考虑到当时已知可用的数据速率，服务器针对候选版本针对其候选版本至少有一部分来计算 尚未发送，如果以可用速率发送的部分开始的任何数量的部分将发生定时错误的最大值。 依赖于计算出的错误与接收缓冲器的当前状态之间的比较来进行用于连续传输的相同或不同版本的选择。 可以预先对存储并稍后检索的用于估计对应于实际传输速率的误差值的传输速率的范围来计算误差值。
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公开(公告)号：US08228386B2
公开(公告)日：2012-07-24
申请号：US11916072
申请日：2006-05-19
CPC分类号： H04N17/004
摘要： Faults resulting in reception of a still, but unknown, frame are recognized by comparing each received frame of the video signal with its predecessor, incrementing a counter in the event that the difference between the frames falls below a threshold; and generating an alarm signal in the event that the count of the counter exceeds a predetermined count. Other types of fault such as loss of signal (i.e. reception of just noise) are recognized by incrementing the counter whenever the difference exceeds a threshold. Similar results may be obtained by instead the monitoring quantization step size and/or number of transmitted bits of a digitally coded signal, and noting that it falls below, or exceeds, a threshold. A preferred option is to compute a complexity measure, being a monotonically increasing function (e.g the product) of the quantization step size and of the number of coded bits and compare this with the threshold value. Faults occasioned by receipt of a particular fixed image instead of the wanted picture are recognized by comparing each frame of the video signal with the fixed image, and generating an alarm signal in the event that the difference between the frames consistently falls below a threshold.
摘要翻译： 通过将视频信号的每个接收帧与其前身进行比较来识别导致静止但未知的帧的接收的故障，在帧之间的差异低于阈值的情况下增加计数器; 并且在计数器的计数超过预定计数的情况下产生报警信号。 当差异超过阈值时，通过增加计数器来识别其它类型的故障，例如信号的丢失（即仅接收噪声）。 可以通过代替监视数字编码信号的发送位的量化步长和/或数量，并注意到其下降或超过阈值来获得相似的结果。 优选的选择是计算复杂性度量，其是量化步长和编码比特数的单调递增函数（例如乘积），并将其与阈值进行比较。 通过将视频信号的每一帧与固定图像进行比较来识别通过接收到特定固定图像而不是想要的图像而引起的故障，并且在帧之间的差异始终低于阈值的情况下产生报警信号。
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公开(公告)号：US08306955B2
公开(公告)日：2012-11-06
申请号：US12920560
申请日：2009-01-19
CPC分类号： H04L67/1095 , H04L65/4076 , H04L67/325
摘要： There is proposed a method of side-loading content onto a mobile device, preferably using a WiFi connection, from a network content store and storing it on the device for viewing later. The method automatically determines when to delete stored content, and more importantly, what content should be transferred over the network in the first instance. The system comprises a content manager which takes into account various parameters to determine what should be transferred and, if required, what should be deleted from the device. The parameters used include user profiles, size of a potential download, as well as available storage space on the device and the predicted duration the device will be in range of a wireless hotspot or broadcast network.
摘要翻译： 提出了一种从网络内容存储器将内容侧向加载到移动设备上，优选地使用WiFi连接的方法，并将其存储在设备上以便稍后查看。 该方法自动确定何时删除存储的内容，更重要的是，首先通过网络传输什么内容。 该系统包括内容管理器，该内容管理器考虑到各种参数以确定应当传送什么，以及如果需要，应该从设备中删除什么。 所使用的参数包括用户简档，潜在下载的大小以及设备上的可用存储空间以及设备将在无线热点或广播网络的范围内的预测持续时间。
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公开(公告)号：US08495698B2
公开(公告)日：2013-07-23
申请号：US12934180
申请日：2009-02-19
IPC分类号： H04N7/16
CPC分类号： H04L65/4084 , H04L65/80 , H04L67/306 , H04N7/163 , H04N7/17318 , H04N21/41407 , H04N21/4402 , H04N21/44222 , H04N21/4424 , H04N21/6581
摘要： A method of intelligently degrading the quality of content, such as video clips and music, stored on a mobile device over time. The method takes into account the probability of a user watching each item of content and either encodes the content at a lower bit-rate, or under certain circumstances, restores the content to its original quality by side-loading the content from a network content store. The bit-rate at which the content is to be encoded is linked to the probability of a piece of content being watched. If the probability of an item of content being watched is high, then the item is stored at a high bit-rate, and conversely, for low probabilities of watching, a low bit-rate is used. If required, in order to restore an item to a higher bit rate, the original high bit rate item can be requested from a network content store. The aim being to store as much content as possible in a given finite memory, but to ensure that content which is likely to be watched is not degraded.
摘要翻译： 随着时间的推移，智能降低存储在移动设备上的内容质量（如视频剪辑和音乐）的方法。 该方法考虑了用户观看每个内容的概率，并且以较低的比特率对内容进行编码，或者在某些情况下，通过从网络内容存储器中的内容进行侧面加载来将内容恢复到其原始质量 。 要对内容进行编码的比特率与正在观看的内容的概率相关联。 如果正在观看的内容项目的概率高，则以高比特率存储该项目，相反地，对于低观看概率，使用低比特率。 如果需要，为了将项目恢复到更高的比特率，可以从网络内容存储器请求原始高比特率项目。 目的是在给定的有限存储器中存储尽可能多的内容，但是确保可能被监视的内容不会降级。
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公开(公告)号：US20080320573A1
公开(公告)日：2008-12-25
申请号：US12278877
申请日：2007-01-11
IPC分类号： H04L9/32
CPC分类号： H04L63/08 , G06F21/31 , H04L63/0876 , H04L67/34
摘要： For users to access network services such as video streaming from a device, users usually have to register themselves with the service first. Most registration mechanisms require a user to input a username, password, date of birth and other details. When implemented through a web interface, this mechanism is relatively easy for the service provider to provision. However, the problem is that users are required to manually enter significant amounts of information which can be both time consuming, especially on a mobile device where there usually no QWERTY input device, and susceptible to unintentional errors. The present invention proposes an automated registration process that does not require a user to enter any details manually except for the initial request to subscribe to a service. The process gathers information automatically about the user and the device used, which is then stored and used for user authentication during subsequent service requests following the initial subscription request. The subsequent requests for service also do not require the user to manually input any user data.
摘要翻译： 对于用户来访问设备的视频流等网络服务，用户通常必须首先注册自己的服务。 大多数注册机制要求用户输入用户名，密码，出生日期和其他细节。 当通过Web界面实现时，该机制对于服务提供商来说相对容易。 然而，问题是用户需要手动输入大量的信息，这些信息既费时又费力，特别是在通常没有QWERTY输入设备的移动设备上，并且容易受到无意的错误的影响。 本发明提出了一种自动注册过程，其不需要用户手动输入任何细节，除了初始请求以订阅服务。 该过程自动收集关于用户和所使用的设备的信息，然后在初始订阅请求之后的后续服务请求期间将其存储并用于用户认证。 随后的服务请求也不需要用户手动输入任何用户数据。
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公开(公告)号：US08218443B2
公开(公告)日：2012-07-10
申请号：US13201268
申请日：2010-02-02
IPC分类号： G01R31/08
CPC分类号： H04L47/10 , H04L45/24 , H04L47/122 , H04L47/193
摘要： There is proposed a modification to the ECN protocol to remove the one-to-one relationship between the network signalling congestion and the sender response to that congestion. The result is to allow a receiver terminal to exhibit some control of bandwidth share relative to other receiver terminals. The idea is to calculate the average CE arrival frequency from received data packets and to set ECE flags at a rate determined as a function of the average CE arrival frequency. Preferably, the function is a multiplier applied to the average CE arrival frequency. The effect of averaging the CE arrival frequency as well as application of a multiplier is a decoupling of the ECE marked ACKs sent by a receiver to the sender, and control of the resulting transmission rate at the sender.
摘要翻译： 提出了对ECN协议的修改以消除网络信令拥塞与对该拥塞的发送者响应之间的一对一关系。 结果是允许接收机终端相对于其他接收机终端表现出对带宽共享的一些控制。 这个想法是从接收的数据包计算平均CE到达频率，并以按照CE平均值到达频率确定的速率来设置ECE标志。 优选地，该功能是适用于平均CE到达频率的乘数。 CE到达频率的平均和乘法器的应用的效果是接收机发送的ECE标记的ACK对发送方的去耦，并且控制发送方产生的传输速率。
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公开(公告)号：US07907529B2
公开(公告)日：2011-03-15
申请号：US12295079
申请日：2007-02-28
申请人： David R Wisely , Rory S Turnbull , Richard Gedge
发明人： David R Wisely , Rory S Turnbull , Richard Gedge
IPC分类号： H04L12/26
CPC分类号： H04L47/724 , H04L45/00 , H04L47/15 , H04L47/19 , H04L47/70 , H04L47/745 , H04L47/801 , H04L47/822
摘要： In order to keep a pathway for time critical data packets (for example for VoIP calls) a dummy flow of data packets is established between a home hub router (20) and a broadband remote access server (25) being the first gateway to the IP network. Should a call be made from the telephone (23), the dummy flow is replaced for the duration of the call by the packets generated for the VoIP call. The dummy flow is adjusted by increasing or decreasing the number of data packets transmitted in order to ensure that a minimum bandwidth having a tolerable delay is available. The dummy flow works by causing non time critical (i.e. P2P data packet flows) to back off and no modification is required to routers and the like incorporated into the network.
摘要翻译： 为了保持时间关键数据分组（例如对于VoIP呼叫）的路径，在家庭中心路由器（20）和作为到IP的第一网关的宽带远程访问服务器（25）之间建立数据分组的虚拟流 网络。 如果通过电话（23）进行呼叫，则通过为VoIP呼叫生成的分组在呼叫期间更换虚拟流。 通过增加或减少发送的数据分组的数量来调整虚拟流，以确保具有可容忍的延迟的最小带宽是可用的。 虚拟流通过使非时间关键（即P2P数据分组流）退出而不起作用，并且不需要对组合到网络中的路由器等进行修改。
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公开(公告)号：US20090116443A1
公开(公告)日：2009-05-07
申请号：US12293892
申请日：2007-03-16
申请人： Matthew D Walker , Nicholas W Farrow , Rory S Turnbull , Nicholas J Sim , Richard J Evenden , Francis J Scahill
发明人： Matthew D Walker , Nicholas W Farrow , Rory S Turnbull , Nicholas J Sim , Richard J Evenden , Francis J Scahill
摘要： This invention relates to maintaining or re-establishing a communications session in a wireless communications system. The present invention provides a method of re-routing a call between a user device and a third party device, the method comprising: setting up a call with the third party device using a first wireless connection between the user device and an access point; monitoring a quality parameter of the first wireless connection at the access point; automatically re-routing the call from the third party device to the user via a second wireless connection between the user device and a base station in response to determining that the quality parameter of the first wireless connection has fallen below a predetermined threshold.
摘要翻译： 本发明涉及在无线通信系统中维护或重新建立通信会话。 本发明提供了一种在用户设备和第三方设备之间重新路由呼叫的方法，所述方法包括：使用所述用户设备和接入点之间的第一无线连接与所述第三方设备建立呼叫; 监视接入点处的第一无线连接的质量参数; 响应于确定第一无线连接的质量参数已经降到预定阈值以下，通过用户设备和基站之间的第二无线连接自动地将呼叫从第三方设备路由到用户。
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