公开(公告)号：US08121115B2

公开(公告)日：2012-02-21

申请号：US11504779

申请日：2006-08-16

申请人： Tomas Frankkila ,  Marten Ericson

发明人： Tomas Frankkila ,  Marten Ericson

IPC分类号： H04L12/56

CPC分类号： H04L47/2416 ,  H04L47/2458 ,  H04L47/564 ,  H04W28/10 ,  H04W72/1221 ,  H04W72/1242

摘要： A packet scheduler reduces or “compresses” the packet transmission delay jitter or delay range where packets experience little or no scheduling delay before transmission. As a result, the number of packets that experience little or no delay is reduced. A preferred example way of compressing the packet transmission delay jitter is to reduce the transmission priority of low delay packets. Compressing the delay jitter is particularly desirable for services like VoIP that require low packet transmission delay jitter.

摘要翻译： 分组调度器减少或“压缩”分组传输延迟抖动或延迟范围，其中分组在传输之前经历很少或没有调度延迟。 结果，减少了经历很少或没有延迟的数据包的数量。 压缩分组传输延迟抖动的优选示例方式是降低低延迟分组的传输优先级。 压缩延迟抖动对于需要低分组传输延迟抖动的VoIP这样的服务是特别需要的。

公开(公告)号：US20080008203A1

公开(公告)日：2008-01-10

申请号：US11504779

申请日：2006-08-16

申请人： Tomas Frankkila ,  Marten Ericson

发明人： Tomas Frankkila ,  Marten Ericson

IPC分类号： H04L12/56 ,  H04L12/54

CPC分类号： H04L47/2416 ,  H04L47/2458 ,  H04L47/564 ,  H04W28/10 ,  H04W72/1221 ,  H04W72/1242

摘要： A packet scheduler reduces or “compresses” the packet transmission delay jitter or delay range where packets experience little or no scheduling delay before transmission. As a result, the number of packets that experience little or no delay is reduced. A preferred example way of compressing the packet transmission delay jitter is to reduce the transmission priority of low delay packets. Compressing the delay jitter is particularly desirable for services like VoIP that require low packet transmission delay jitter.

摘要翻译： 分组调度器减少或“压缩”分组传输延迟抖动或延迟范围，其中分组在传输之前经历很少或没有调度延迟。 结果，减少了经历很少或没有延迟的数据包的数量。 压缩分组传输延迟抖动的优选示例方式是降低低延迟分组的传输优先级。 压缩延迟抖动对于需要低分组传输延迟抖动的VoIP这样的服务是特别需要的。

公开(公告)号：US09509805B2

公开(公告)日：2016-11-29

申请号：US14361797

申请日：2011-12-01

申请人： Ingemar Johansson ,  Tomas Frankkila

发明人： Ingemar Johansson ,  Tomas Frankkila

IPC分类号： H04L12/28 ,  G01R31/08 ,  H04L29/06 ,  H04L12/811 ,  H04L12/801

CPC分类号： H04L69/04 ,  H04L47/33 ,  H04L47/38 ,  H04L69/22 ,  H04L69/24

摘要： An ECN flow controller (22) reduces packet header compression overhead due to high ECN rate. It includes an ECN filter (24) configured to redistribute, with at least approximately maintained ECN rate, ECN-CE marks among headers to reduce switching between ECN-CE marked and ECT marked headers.

摘要翻译： ECN流量控制器（22）由于高ECN速率而降低了包头压缩开销。 它包括一个ECN滤波器（24），配置为在标题中以至少近似维持的ECN速率重新分配ECN-CE标记，以减少ECN-CE标记和ECT标记的标题之间的切换。

公开(公告)号：US07864814B2

公开(公告)日：2011-01-04

申请号：US12092884

申请日：2005-11-07

申请人： Ingemar Johansson ,  Tomas Frankkila

发明人： Ingemar Johansson ,  Tomas Frankkila

IPC分类号： H04J3/06

CPC分类号： H04J3/0632

摘要： A control logic means preferably for a receiver comprising a jitter buffer means adapted to receive and buffer incoming frames or packets and to extract data frames from the received packets, a decoder connected to the jitter buffer means adapted to decode the extracted data frames, and a time scaling means connected to the decoder adapted to play out decoded speech frames adaptively. The control logic means comprises knowledge of whether a state recovery function is available and is adapted to retrieve at least one parameter from at least one of the jitter buffer means, the time scaling means, and the decoder, to adaptively control at least one of an initial buffering time of said jitter buffer means, the knowledge of the availability of the state recovery function, and a time scaling amount of said time scaling means from the time scaling means or the decoder.

摘要翻译： 控制逻辑优选地适用于包括适于接收和缓冲输入帧或分组并从接收到的分组中提取数据帧的抖动缓冲器装置的接收机，连接到抖动缓冲器装置的解码器，其适于对所提取的数据帧进行解码，以及 连接到解码器的时间缩放装置，适于自适应地播放解码的语音帧。 控制逻辑装置包括有关状态恢复功能是否可用并且适于从抖动缓冲器装置，时间缩放装置和解码器中的至少一个检索至少一个参数的知识，以自适应地控制以下各项中的至少一个： 所述抖动缓冲器的初始缓冲时间意味着，状态恢复功能的可用性的知识，以及来自时间缩放装置或解码器的所述时间缩放装置的时间缩放量。

公开(公告)号：US09307079B2

公开(公告)日：2016-04-05

申请号：US12518222

申请日：2007-11-30

申请人： Tomas Frankkila ,  Daniel Enström ,  Ingemar Johansson

发明人： Tomas Frankkila ,  Daniel Enström ,  Ingemar Johansson

IPC分类号： G06F15/16 ,  H04M3/487 ,  H04M7/00

CPC分类号： H04M3/487 ,  H04M7/006 ,  H04M2203/352

摘要： In order to efficiently handle the switch between user media and announcement media, a basic step is to first determine a configuration of the user media. Next, a configuration of the announcement media to be presented is determined based on the determined user media configuration. Subsequently, the announcement media is configured according to the announcement media configuration, and the configured announcement media is finally sent to the intended user. In this way, the overall appearance or sound of the announcement will be virtually the same as or at least similar to the overall appearance or sound of the user media, preferably without distortions. This allows the user to perceive the announcement as clearly as possible.

摘要翻译： 为了有效地处理用户媒体和通告媒体之间的切换，基本步骤是首先确定用户媒体的配置。 接下来，基于所确定的用户媒体配置来确定要呈现的通知媒体的配置。 随后，根据通知媒体配置配置通知媒体，并且配置的通知媒体最终被发送到预期用户。 以这种方式，通知的总体外观或声音将几乎与用户媒体的整体外观或声音相似或至少相似，优选地没有扭曲。 这允许用户尽可能清楚地看到该通知。

公开(公告)号：US20130250796A1

公开(公告)日：2013-09-26

申请号：US13989133

申请日：2010-11-30

申请人： Tomas Frankkila ,  Yang Zuo

发明人： Tomas Frankkila ,  Yang Zuo

IPC分类号： H04W24/10

CPC分类号： H04W24/10 ,  H04L47/14 ,  H04L47/2416 ,  H04L47/26 ,  H04L47/28 ,  H04L47/283 ,  H04L47/36

摘要： A method and arrangement for employing media layer adaptation in a wireless communication of media in data packets from a sending node to a receiving node, by determining a fitting frame aggregation scheme in an effective and accurate manner. An arrival time AT and generation time GT are monitored for packets when received at the packet receiving node. A difference ATdiff in the arrival time of consecutive packets and a difference GTdiff in the generation time of the packets, are calculated. Then, an inter-arrival measure IA is calculated as the deviation between the arrival time difference ATdiff and generation time difference GTdiff. When the inter-arrival jitter exceeds a preset threshold (Th), a frame aggregation scheme is determined based on the calculated inter-arrival jitter IA and applied in the packet communication.

摘要翻译： 通过以有效和准确的方式确定拟合帧聚合方案，在从发送节点到接收节点的数据分组中的介质的无线通信中采用媒体层适配的方法和装置。 在分组接收节点处接收到分组时，监视到达时间AT和生成时间GT。 计算连续数据包到达时间的差值ATdiff和数据包生成时间差值GTdiff。 然后，计算到达时间差值ATdiff与产生时间差值GTdiff之间的偏差。 当到达之间的抖动超过预设阈值（Th）时，基于所计算的到达之间的抖动IA并在分组通信中应用帧聚合方案。

公开(公告)号：US20090245272A1

公开(公告)日：2009-10-01

申请号：US12278493

申请日：2007-02-06

申请人： Janne Peisa ,  Tomas Frankkila ,  Ingemar Johansson ,  Daniel Enstrom ,  Mats Sagfors

发明人： Janne Peisa ,  Tomas Frankkila ,  Ingemar Johansson ,  Daniel Enstrom ,  Mats Sagfors

IPC分类号： H04L12/54

CPC分类号： H04L1/0014 ,  H04L1/1812

摘要： In a method of improved media frame transmission in a communication network. Initially a plurality of “original” or regular media frames are provided for transmission. According to the invention, robust representations of the provided regular media frames are generated and stored locally. Subsequently, one or more of the regular media frames is/are transmitted. The invention detects an indication of a loss of a transmitted media frame, and the idea is to transmit, in response to a detected frame loss, a stored robust representation of the lost media frame and/or a stored robust representation of a subsequent, not yet transmitted, media frame to increase the media quality.

摘要翻译： 在通信网络中改进媒体帧传输的方法中。 最初提供多个“原始”或常规媒体帧用于传输。 根据本发明，提供的常规媒体帧的鲁棒表示被生成并存储在本地。 随后，传送一个或多个常规媒体帧。 本发明检测到传输的媒体帧丢失的指示，并且该思想是响应于检测到的帧丢失而发送丢失的媒体帧的存储的鲁棒表示和/或后续的不存在的鲁棒表示 传播媒体框架，提高媒体品质。

公开(公告)号：US20090103450A1

公开(公告)日：2009-04-23

申请号：US12278486

申请日：2007-01-30

申请人： Janne Peisa ,  Mats Sagfors ,  Tomas Frankkila ,  Daniel Enstrom ,  Ghyslain Pelletier

发明人： Janne Peisa ,  Mats Sagfors ,  Tomas Frankkila ,  Daniel Enstrom ,  Ghyslain Pelletier

IPC分类号： G06F11/30 ,  H04B7/216

CPC分类号： H04L1/0002 ,  H04L1/0017 ,  H04L1/0025 ,  H04L1/0038 ,  H04L1/1812 ,  H04L1/1877 ,  H04L5/0007 ,  H04L5/0037 ,  H04L5/006 ,  H04L5/0064 ,  H04W72/1242 ,  H04W72/1268

摘要： User equipment in a wireless communication system monitors scheduling information and locally detects a change in link data rate of the uplink channel based on the monitored scheduling information. In this way, a change in link data rate can be detected directly without significant delay. This direct or early detection of a rate change is then combined with an appropriate system reaction. The information of the detected change in link data rate is preferably utilized for adapting the application data rate of an IP application running in the user equipment. As an alternative, or as a complement, data packets are classified based on relative importance and selected for transfer of information over the uplink channel based on the classification of data packets and in dependence on the detected change in link data rate.

摘要翻译： 无线通信系统中的用户设备基于所监视的调度信息来监视调度信息并本地检测上行链路信道的链路数据速率的变化。 以这种方式，可以直接检测链路数据速率的变化而没有显着的延迟。 然后将这种速率变化的直接或早期检测与适当的系统反应组合。 所检测到的链接数据速率变化的信息优选地用于调整在用户设备中运行的IP应用的应用数据速率。 作为替代或补充，基于相对重要性对数据分组进行分类，并且基于数据分组的分类并根据检测到的链路数据速率的变化，通过上行链路信道来选择信息的传输。

公开(公告)号：US20050227657A1

公开(公告)日：2005-10-13

申请号：US10819376

申请日：2004-04-07

申请人： Tomas Frankkila ,  Jonas Svedberg ,  Krister Svanbro ,  Bjorn Svensson ,  Tomas Jonsson

发明人： Tomas Frankkila ,  Jonas Svedberg ,  Krister Svanbro ,  Bjorn Svensson ,  Tomas Jonsson

IPC分类号： G10L13/06 ,  H04L12/56 ,  H04Q7/38

CPC分类号： G10L21/04

摘要： Perceived interactivity in user communications is achieved by reducing a perceived delay switching the active transmitter in the communication without having to reduce actual transmission and setup delays associated with a communication exchange. A sound signal is identified in the user communication. The sound signal is analyzed to identify or estimate a sound signal segment. The sound signal segment is preferably (though not necessarily) located at the beginning or the end of the sound signal. The sound signal segment may be selected directly from the sound signal itself, from a modified version of the sound signal, or from a signal associated with the sound signal. A determination is made that a length or duration of the sound signal segment should be or can be modified. One or more modifications for the sound signal segment are determined and are provided to one or more processing units to perform the modification(s).

摘要翻译： 用户通信中的感知交互性通过减少在通信中切换有源发射机的感知延迟而不用减少与通信交换相关联的实际传输和建立延迟来实现。 在用户通信中识别声音信号。 分析声音信号以识别或估计声音信号段。 声音信号段优选地（尽管不一定）位于声音信号的开始或结束处。 可以从声音信号本身，声音信号的修改版本或与声音信号相关联的信号直接选择声音信号段。 确定声音信号段的长度或持续时间应该是或可以被修改。 确定声音信号段的一个或多个修改，并将其提供给一个或多个处理单元以执行修改。

公开(公告)号：US06873954B1

公开(公告)日：2005-03-29

申请号：US09655326

申请日：2000-09-05

申请人： Jim Sundqvist ,  Tomas Frankkila ,  Anders Nohlgren

发明人： Jim Sundqvist ,  Tomas Frankkila ,  Anders Nohlgren

IPC分类号： H04J3/06 ,  G10L13/02

CPC分类号： H04J3/0632

摘要： Audio artifacts due to overrun or underrun in a playout buffer caused by the sampling rates at a sending and receiving side not being at the same rate are reduced. An LPC-residual is modified on a sample-by-sample basis. The LPC-residual block, which includes N samples, is converted to a block comprising N+1 or N−1 samples. A sample rate controller decides whether samples should be added to or removed from the LPC-residual. The exact position at which to add respective remove samples is either chosen arbitrarily or found by searching for low energy segments in the LPC-residual. A speech synthesiser module then reproduces the speech. By using the proposed sample rate conversion method the playout buffer can be continuously controlled. Furthermore, since the method works on a sample-by-sample basis the buffer can be kept to a minimum and hence no extra delay is introduced.

摘要翻译： 由于发送和接收侧的采样率不在同一速率引起的播放缓冲器中的超载或欠载的音频伪影减少。 LPC-残差在逐个样本的基础上修改。 包括N个采样的LPC残差块被转换为包括N + 1或N-1个采样的块。 采样率控制器决定是否应将样本添加到LPC残差中或从LPC残差中去除。 添加相应去除样本的确切位置是任意选择的或通过搜索LPC残差中的低能量段来找到。 语音合成器模块然后再现语音。 通过使用提出的采样率转换方法，可以连续控制播放缓冲器。 此外，由于该方法在逐个采样的基础上工作，因此缓冲器可以保持最小，因此不会引入额外的延迟。