公开(公告)号：EP1845519A2

公开(公告)日：2007-10-17

申请号：EP07109801

申请日：2004-12-15

Applicant: ERICSSON TELEFON AB L M

Inventor： JOHANSSON INGEMAR ,  TALEB ANISSE ,  BRUHN STEFAN ,  ENSTROEM DANIEL

IPC: G10L19/008 ,  H04H20060101 ,  H04S3/00

CPC classification number: G10L19/008

Abstract: A method of encoding multi-channel audio signals comprises generating of a first output signal (x' mono ), being encoding (38) parameters representing a main signal (x mono ). The main signal (x mono ) is a first linear combination (34) of signals (16A,16B) of at least a first and a second channel. The method further comprises generating (30) of a second output signal (p side ), being encoding parameters representing a side signal (x side ). The side signal (x side ) is a second linear combination (36) of signals (16A,16B) of at least the first and the second channel within an encoding frame. The method is characterised in that the generating of the second output signal further comprises scaling of the side signal (x side ) to an energy contour of the main signal (x mono ). A method of decoding is also presented as well as an encoder, a decoder and audio system, all according to the same basic idea.

Abstract translation: 编码多声道音频信号的方法包括：一个第一输出信号（X“单声道）被编码的产生（38）表示的主信号（X单声道）的参数。 主信号（X单声道）是至少第一和第二信道的信号（16A，16B）的第一线性组合（34）。 是代表一个侧信号（X侧）的编码参数的第二输出信号（P侧）的方法，还包括：生成（30）。 侧信号（X侧）是至少为所述第一和在编码帧内的第二信道的信号（16A，16B）的第二线性组合（36）。 该方法DASS侧信号（X侧）的第二输出信号还包括缩放的产生对主信号（X单声道）的能量轮廓。 因此解码的方法是在编码器呈现以及，解码器和音频系统中，所有gemäß到相同的基本思想。

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：WO2007102780A3

公开(公告)日：2007-11-15

申请号：PCT/SE2007050130

申请日：2007-03-07

Applicant: ERICSSON TELEFON AB L M ,  JOHANSSON INGEMAR ,  PELLETIER GHYSLAIN

Inventor： JOHANSSON INGEMAR ,  PELLETIER GHYSLAIN

IPC: H04L29/06 ,  H04L12/56

CPC classification number: H04L69/04 ,  H04L49/00 ,  H04L69/22

Abstract: A method and a communication station (5) for determining a packet format of a data packet based on at least one compressed header information field. A header packet format of a data packet is determined based on the determined packet format of a partially or completely compressed header information part. The station may also determine a codec rate (10) based on the result of a previously performed compression of at least one header information field. In one embodiment, internal cross-layer signaling from a header generator module (6) to a payload generator module (7) in the communication station is used for signaling information associated with determining the header packet format of the data packet. The method reduces bandwidth fluctuations associated with services such as text messaging, audio, or audiovisual services; lowers the delay variations and the erasure rates for application media streams; and provides for faster set-up of sessions.

Abstract translation: 一种用于基于至少一个压缩报头信息字段来确定数据分组的分组格式的方法和通信站（5）。 基于所确定的部分或完全压缩的报头信息部分的分组格式来确定数据分组的报头分组格式。 基站还可以基于先前执行的至少一个报头信息字段的压缩的结果来确定编解码器速率（10）。 在一个实施例中，来自报头发生器模块（6）到通信站中的有效载荷发生器模块（7）的内部跨层信令用于与确定数据分组的报头分组格式相关联的信令信息。 该方法减少与诸如文本消息，音频或视听服务之类的服务相关联的带宽波动; 降低应用媒体流的延迟变化和擦除率; 并提供更快的会话设置。

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：WO0031719A3

公开(公告)日：2003-03-20

申请号：PCT/SE9902023

申请日：1999-11-08

Applicant: ERICSSON TELEFON AB L M

Inventor： EKUDDEN ERIK ,  HAGEN ROAR ,  JOHANSSON INGEMAR

IPC: G10L19/00 ,  H04B14/04 ,  H04M1/00 ,  H04M1/725

CPC classification number: G10L19/012

Abstract: The quality of comfort noise generated by a speech decoder (93) during non-speech periods is improved by modifying (30, 75) comfort noise parameter values (33) normally used to generate the comfort noise. The comfort noise parameter values are modified in response to variability information (43) associated with a background noise parameter. The modified comfort noise parameter values (35) are then used to generate the comfort noise.

Abstract translation: 通过修改（30,75）通常用于产生舒适噪声的舒适噪声参数值（33）来提高在非语音周期期间由语音解码器（93）产生的舒适噪声的质量。 响应于与背景噪声参数相关联的可变性信息（43）来修改舒适噪声参数值。 改进的舒适噪声参数值（35）然后用于产生舒适噪声。

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：WO0243271A3

公开(公告)日：2002-08-22

申请号：PCT/SE0102531

申请日：2001-11-13

Applicant: ERICSSON TELEFON AB L M

Inventor： JOHANSSON INGEMAR

IPC: H04B7/005 ,  H04L7/04 ,  H04L25/02 ,  H04B7/26

CPC classification number: H04B7/005 ,  H04L7/042 ,  H04L25/0242

Abstract: A method and apparatus for acquiring time synchronization, as well as a channel estimate and a noise estimate for a received signal are described. First, a burst of data which includes a training sequence is received. Then, groups of received training sequence measurements are created. For each of these groups, a set of linear equations is set up. Next, methods according to the present invention determine the least square estimate by solving the matrix equation H = (A A) A L for each of the groups. This least square estimate is performed in several steps which reduce the number of calculations to be performed. The term A A is the same for each group and can be computed once prior to computing the least square estimates. The A L terms are calculated separately for each group. In the process of calculating the A L term for each group, the A L term calculated for the previous group can be reused to reduce the total number of calculations. Once the A A and A L terms are calculated, then the least square estimate for each group is determined. The least square estimate is then used to determine a noise estimate for each group. Again, mechanisms for reuse of certain calculations associated with the determination of the noise estimate that reduces the computational complexity associated therewith. Once the noise estimate is calculated for each group, the method determines which of the groups has the lowest valued noise estimate. That group, according to one exemplary embodiment, is chosen as the synchronization group. As an alternative, the group which has the greatest estimated signal-to-noise ratio can be selected as the synchronization group in this step. The least square estimate of the synchronization group is then identified as the channel estimate.

Abstract translation: 描述了用于获取时间同步的方法和设备，以及用于接收信号的信道估计和噪声估计。 首先，接收包括训练序列的数据突发。 然后，创建接收到的训练序列测量组。 对于这些组中的每一组，建立一组线性方程。 接下来，根据本发明的方法通过求解每个组的矩阵方程来确定最小二乘估计。 这个最小二乘估计是在几个步骤中执行的，这减少了要执行的计算次数。 术语A A对于每个组是相同的，并且可以在计算最小二乘估计之前计算一次。 A LT项对于每个组分别计算。 在计算每个组的A L项的过程中，可以重新使用为前一组计算的A 项，以减少总计算次数。 一旦计算了A T和A T项，就确定了每个组的最小二乘估计。 然后使用最小二乘估计来确定每个组的噪声估计。 同样，重新使用与确定噪声估计相关联的某些计算的机制降低了与之相关的计算复杂度。 一旦为每个组计算了噪声估计，该方法就确定哪个组具有最低的噪声估计值。 根据一个示例性实施例，该组被选择为同步组。 也可以选择具有最大估计信噪比的组作为本步骤中的同步组。 随后将同步组的最小二乘估计识别为信道估计。

公开(公告)号：ES2385749T3

公开(公告)日：2012-07-31

申请号：ES07118035

申请日：2007-10-08

Applicant: ERICSSON TELEFON AB L M ,  ERICSSON TELEFON AB L M

Inventor： LUDWIG REINER ,  ENSTROEM DANIEL ,  JOHANSSON INGEMAR ,  HEDBERG TOMAS

IPC: H04L12/56 ,  H04L47/31

Abstract: Un método para controlar un receptor de una comunicación mediante unidades de datos desde una fuente dedatos, estando el citado receptor asociado con un primer terminal de una red de comunicación mediante unidadesde datos que proporciona al primer terminal un servicio de transporte caracterizado por un conjunto de valoresasociados con respectivos parámetros de calidad de servicio, comprendiendo los citados parámetros de calidad deservicio un primer parámetro de capacidad de transmisión mínima, estando la citada fuente de datos asociada conun segundo terminal, comprendiendo el citado método:- determinar (S40), por parte del receptor, si una unidad de datos recibida contiene una marca de indicaciónde congestión,- si se detecta la citada marca de indicación de congestión, enviar (S42) desde el receptor a la citada fuentede datos un mensaje para ajustar una tasa de transmisión de datos de la citada fuente de datos, indicando elcitado mensaje como límite superior para la citada tasa de transmisión de datos un valor correspondiente a unvalor asociado con el primer parámetro de capacidad de transmisión mínima para el citado servicio detransporte proporcionado al citado primer terminal asociado con el citado receptor.

公开(公告)号：WO2007073328A2

公开(公告)日：2007-06-28

申请号：PCT/SE2006050560

申请日：2006-12-07

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE ,  JANSSON FREDRIK ,  HAAKANSSON STEFAN ,  JOHANSSON INGEMAR ,  BRUHN STEFAN ,  KIM MOO YOUNG

Inventor： TALEB ANISSE ,  JANSSON FREDRIK ,  HAAKANSSON STEFAN ,  JOHANSSON INGEMAR ,  BRUHN STEFAN ,  KIM MOO YOUNG

IPC: H04L12/66 ,  H04L1/18 ,  H04L12/56

CPC classification number: H04L1/0041 ,  H04L1/0009 ,  H04L1/0026 ,  H04L1/18 ,  H04L49/552 ,  H04L2001/0097

Abstract: A network processing node (e.g., MGW, MRFP) and method are described herein that can: (1) receive packets on a first heterogeneous link (e.g., wireless link); (2) manipulate the received packets based on known characteristics about a second heterogeneous link (e.g., "Internet" link); and (3) send the manipulated packets on the second heterogeneous link (e.g., "Internet" link). For example, the network processing node can manipulate the received packets by adding redundancy, removing redundancy, frame aggregating (re-packetizing), recovering lost packets and/or re-transmitting packets.

Abstract translation: 这里描述的网络处理节点（例如，MGW，MRFP）和方法可以：（1）在第一异构链路（例如，无线链路）上接收分组; （2）基于关于第二异构链路（例如，“因特网”链路）的已知特征来操纵接收到的分组; 和（3）在第二异构链路（例如，“互联网”链路）上发送操纵的分组。 例如，网络处理节点可以通过增加冗余，去除冗余，帧聚合（重新打包），恢复丢失的分组和/或重新发送分组来操纵接收到的分组。

公开(公告)号：WO0207323A3

公开(公告)日：2002-09-06

申请号：PCT/SE0101493

申请日：2001-06-28

Applicant: ERICSSON TELEFON AB L M

Inventor： HAAKANSSON STEFAN ,  JOHANSSON INGEMAR ,  SVANBRO KRISTER ,  JONSSON LARS-ERIK ,  DEGERMARK MIKAEL ,  HANNU HANS

IPC: H04L1/00 ,  H03M7/30 ,  H03M13/09 ,  H04L12/56 ,  H04L29/06 ,  H04Q7/38

CPC classification number: H03M13/6312 ,  H03M7/30 ,  H03M13/09 ,  H03M13/098 ,  H04L69/04 ,  H04L69/22 ,  Y10S370/913

Abstract: In packet communications that employ header compression/decompression, the computational complexity of checksum generation can be reduced by re-using static checksum information associated with header bits (S) that do not change from header to header. The static checksum information can be used together with information about header bits (T) that do change from header to header, in order to generate a desired checksum (CS). The checksum can then be used to verify a reconstructed header (17) produced from a compressed header by a header decompressor.

Abstract translation: 在采用报头压缩/解压缩的分组通信中，可以通过重新使用与从报头到报头不改变的报头比特（S）相关联的静态校验和信息来减少校验和生成的计算复杂度。 静态校验和信息可以与关于从头到头改变的头比特（T）的信息一起使用，以便生成期望的校验和（CS）。 然后可以使用校验和来验证由标题解压缩器从压缩报头产生的重建报头（17）。