公开(公告)号：WO2008069723A2

公开(公告)日：2008-06-12

申请号：PCT/SE2007001061

申请日：2007-11-30

Applicant: ERICSSON TELEFON AB L M ,  FRANKKILA TOMAS ,  ENSTROEM DANIEL ,  JOHANSSON INGEMAR

Inventor： FRANKKILA TOMAS ,  ENSTROEM DANIEL ,  JOHANSSON INGEMAR

IPC: H04M3/487

CPC classification number: H04M3/487 ,  H04M7/006 ,  H04M2203/352

Abstract: In order to efficiently handle the switch between user media and announcement media, a basic step (Sl) is to first determine a configuration of the user media. Next, a configuration of the announcement media to be presented is determined (S2) based on the determined user media configuration. Subsequently, the announcement media is configured (S3) according to the announcement media configuration, and the configured announcement media is finally sent (S4) 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.

Abstract translation: 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（S1）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告媒体的配置（S2）。 随后，根据通告媒体配置配置通告媒体（S3），并且配置的通告媒体最终被发送（S4）给目标用户。 以这种方式，通告的整体外观或声音将实质上与用户媒体的整体外观或声音相同或至少相似，优选没有失真。 这允许用户尽可能清楚地感知通告。

公开(公告)号：WO2008008025A3

公开(公告)日：2008-03-06

申请号：PCT/SE2007050377

申请日：2007-05-30

Applicant: ERICSSON TELEFON AB L M

Inventor： FRANKKILA TOMAS ,  WAENSTEDT STEFAN ,  ERICSON MAARTEN

IPC: H04L12/56 ,  H04W72/12

CPC classification number: H04L47/2416 ,  H04L47/2458 ,  H04L47/564 ,  H04W28/10 ,  H04W72/1221 ,  H04W72/1242

Abstract: 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.

Abstract translation: 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：WO2008069722A3

公开(公告)日：2008-07-24

申请号：PCT/SE2007001050

申请日：2007-11-28

Applicant: ERICSSON TELEFON AB L M ,  JOHANSSON INGEMAR ,  ENSTROEM DANIEL ,  FRANKKILA TOMAS

Inventor： JOHANSSON INGEMAR ,  ENSTROEM DANIEL ,  FRANKKILA TOMAS

IPC: H04N7/00

CPC classification number: H04L65/604 ,  G10L19/005 ,  G10L19/24 ,  H04L65/1089 ,  H04L65/605 ,  H04L65/80

Abstract: A receiver (100) includes a detector (140) for detecting a change in source of incoming media during an on-going communication session, and means to provide a reset signal in order to reset decoder states of a decoder (120) in response to such a detected change before decoding new incoming media. In this way, a state mismatch can be avoided without the need for several active decoder instances in the receiver (100), leading to substantial savings with respect to overall complexity, memory usage and power consumption. This also means that media distortions can be eliminated or at least reduced when the decoded media is finally rendered by a player (130).

Abstract translation: 接收器（100）包括检测器（140），用于检测正在进行的通信会话期间进入的媒体的源的变化，以及提供复位信号的装置，以便响应于解码器（120）的解码器状态来重置 这种检测到的变化在解码新的传入媒体之前。 以这种方式，可以避免在接收器（100）中不需要几个主动解码器实例的状态失配，导致相对于整体复杂性，存储器使用和功耗的显着节省。 这也意味着当解码的媒体最终由玩家（130）呈现时，可以消除或至少减少媒体失真。

公开(公告)号：WO0118790A8

公开(公告)日：2002-02-28

申请号：PCT/SE0001687

申请日：2000-09-01

Applicant: ERICSSON TELEFON AB L M

Inventor： SUNDQVIST JIM ,  FRANKKILA TOMAS ,  NOHLGREN ANDERS

IPC: H04J3/06 ,  G10L13/00

CPC classification number: H04J3/0632

Abstract: The present invention relates to methods for improving speech quality in e.g. an IP-telephony system. The invention reduces audio artefacts being 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. The inventive solution modifies an LPC-residual on a sample-by-sample basis. The LPC-residual block comprising N samples is converted to a block comprising N+1 or N-1 samples. A sample rate controller 400 decides whether samples should be added to or removed from the LPC-residual. The exact position where 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 430 then reproduces the speech. By using the proposed sample rate conversion method the playout buffer 440 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.

Abstract translation: 本发明涉及用于提高语音质量的方法。 一个IP电话系统。 本发明减少了由于发送和接收侧的采样速率不是相同速率而导致的播放缓冲器中的超载或欠载引起的音频伪影。 本发明的解决方案在逐个样本的基础上修改LPC残差。 包括N个样本的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400确定是否应该将样本添加到LPC残差中或从LPC残差中去除。 添加相应删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430然后再现语音。 通过使用所提出的采样率转换方法，可以连续地控制播出缓冲器440。 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。

公开(公告)号：WO2009056968A2

公开(公告)日：2009-05-07

申请号：PCT/IB2008002926

申请日：2008-10-31

Applicant: ERICSSON TELEFON AB L M ,  RAMESH RAJARAM ,  BALACHANDRAN KUMAR ,  FRANKKILA TOMAS

Inventor： RAMESH RAJARAM ,  BALACHANDRAN KUMAR ,  FRANKKILA TOMAS

IPC: H04L29/06

CPC classification number: H04L65/605 ,  H04L65/103 ,  H04L65/80

Abstract: The present invention sends multiple versions of a multimedia packet to the base station, and, based on the radio channel and traffic characteristics, an appropriate version of the multimedia packet is selected to send to the mobile station at a given time. In this way, source transmission is improved to instantaneous conditions. The steps of the present invention are performed in conjunction with RTP used for multimedia transmission over internet protocol (IP) networks. In a first embodiment, the multiple versions are sent to the base station in the same RTP packet, and the base station strips out the extraneous versions. In a second embodiment, the base station receives multiple RTP packets having identical information in the packet header in many fields, and selects an appropriate one among these for transmission to the mobile station, discarding the rest.

Abstract translation: 本发明向基站发送多个版本的多媒体分组，并且基于无线电信道和业务特性，选择适当版本的多媒体分组以在给定时间向移动台发送。 以这种方式，源传输被改进到瞬时条件。 本发明的步骤与用于通过因特网协议（IP）网络的多媒体传输的RTP一起执行。 在第一实施例中，多个版本在相同的RTP分组中发送到基站，并且基站剥离外部版本。 在第二实施例中，基站在多个字段中在分组报头中接收到具有相同信息的多个RTP分组，并且选择其中适当的一个用于传输到移动台，丢弃其余的分组。

公开(公告)号：WO2007091968A3

公开(公告)日：2007-10-04

申请号：PCT/SE2007050071

申请日：2007-02-06

Applicant: ERICSSON TELEFON AB L M ,  PEISA JANNE ,  SAAGFORS MATS ,  ENSTROEM DANIEL ,  JOHANSSON INGEMAR ,  FRANKKILA TOMAS

Inventor： PEISA JANNE ,  SAAGFORS MATS ,  ENSTROEM DANIEL ,  JOHANSSON INGEMAR ,  FRANKKILA TOMAS

IPC: G10L19/00 ,  H04L1/18

CPC classification number: H04L1/1819 ,  H04L1/0014 ,  H04L1/1877

Abstract: 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.

Abstract translation: 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：ES2615877T3

公开(公告)日：2017-06-08

申请号：ES13742291

申请日：2013-06-25

Applicant: ERICSSON TELEFON AB L M (PUBL)

Inventor： GABIN FREDERIC ,  ERIKSSON ANDERS K ,  FRANKKILA TOMAS

IPC: H04L29/06 ,  H04M1/19

Abstract: Método, llevado a cabo por un primer nodo (110), para gestionar el procesado o tratamiento de un flujo continuo de audio, en donde el flujo continuo de audio es recibible de un segundo nodo (120), en donde el método comprende: enviar (204), al segundo nodo (120), información referente a por lo menos una preferencia relativa a características acústicas del flujo continuo de audio, en donde la por lo menos una preferencia relativa a características acústicas se refiere a uno o más de: un nivel de supresión de ruido en el flujo continuo de audio, un nivel de voz en el flujo continuo de audio, y un ancho de banda del flujo continuo de audio, en donde la por lo menos una preferencia relativa a características acústicas es determinada por un usuario previo o actual del primer nodo (110), o en donde la por lo menos una preferencia relativa a características acústicas es determinada por un operador de un sistema (100) de radiocomunicaciones que comprende el primer y el segundo nodos (110, 120); y recibir (210) el flujo continuo de audio del segundo nodo (120), en donde el flujo continuo de audio ha sido tratado, por el segundo nodo (120), como respuesta a la información referente a la por lo menos una preferencia relativa a las características acústicas del flujo continuo de audio.

公开(公告)号：PL2163069T3

公开(公告)日：2011-04-29

申请号：PL07748190

申请日：2007-05-30

Applicant: ERICSSON TELEFON AB L M

Inventor： ENSTRÖM DANIEL ,  FRANKKILA TOMAS

IPC: H04L49/9023

公开(公告)号：CA2656859C

公开(公告)日：2016-07-26

申请号：CA2656859

申请日：2007-05-30

Applicant: ERICSSON TELEFON AB L M

Inventor： FRANKKILA TOMAS ,  WAENSTEDT STEFAN ,  ERICSON MARTEN

IPC: H04L47/56 ,  H04L12/66 ,  H04L47/80 ,  H04W72/12

Abstract: 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.

公开(公告)号：BRPI1015160A2

公开(公告)日：2016-04-19

申请号：BRPI1015160

申请日：2010-04-06

Applicant: ERICSSON TELEFON AB L M

Inventor： ENSTRÖM DANIEL ,  BRUHN STEFAN ,  FRANKKILA TOMAS

IPC: H04L29/06 ,  H04W28/18 ,  H04W76/02