公开(公告)号：WO2009130042A3

公开(公告)日：2010-07-22

申请号：PCT/EP2009003014

申请日：2009-04-24

Applicant: ERICSSON TELEFON AB L M ,  KELLER RALF ,  RANKE KARL-PETER ,  NOHLGREN ANDERS

Inventor： KELLER RALF ,  RANKE KARL-PETER ,  NOHLGREN ANDERS

IPC: H04W76/02

CPC classification number: H04W76/021 ,  H04L65/1016 ,  H04L65/1083 ,  H04W36/0033 ,  H04W80/10

Abstract: The invention refers to supporting a communication established between a first mobile terminal (18) towards a telecommunications network (1), the telecommunications network (1) comprising a first radio access network (13) and a second radio access network (14) and an application server (10), wherein the application server performs the steps of receiving an identifier that a session setup was initiated with respect to the first mobile terminal (18), wherein the identifier is adapted to identify the session, entering the identifier into an identifier list comprising one or a plurality of a identifiers for identifying sessions with respect to the first mobile terminal (18), receiving an indication of a session transfer form the first radio access network (13) to the second radio access network (14) or vice versa, and providing the identifier list in response to the indication. The invention further refers to an application server (10) and corresponding computer program.

Abstract translation: 本发明涉及支持在朝向电信网络（1）的第一移动终端（18）之间建立的通信，所述电信网络（1）包括第一无线电接入网络（13）和第二无线电接入网络（14），以及 应用服务器（10），其中所述应用服务器执行以下步骤：接收关于所述第一移动终端（18）发起会话建立的标识符，其中所述标识符适于识别所述会话，将所述标识符输入到标识符 列表，其包括用于识别关于第一移动终端（18）的会话的一个或多个标识符，从第一无线电接入网络（13）接收到第二无线电接入网络（14）的会话传送的指示或副 并且响应于指示提供标识符列表。 本发明还涉及应用服务器（10）和相应的计算机程序。

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：DE60129327D1

公开(公告)日：2007-08-23

申请号：DE60129327

申请日：2001-05-22

Applicant: ERICSSON TELEFON AB L M

Inventor： SUNDQVIST JIM ,  LENNESTAL HAKAN ,  NOHLGREN ANDERS ,  LINDQVIST MORGAN ,  WANG WEI

IPC: H04L12/64

Abstract: The present invention relates to a receiver system in a communication system supporting packet-based communicatior (e.g., an IP-network), including a receiver, speech decoder ( 40 ) and a jitter buffer ( 20 ) for handling delay variations in the reception of a speech signal consisting of packets containing frames with encoded speech. A jitter buffer controller ( 50 ) is provided for keeping information about the functional size of the jitter buffer ( 20 ) and for providing the speech decoder ( 40 ) with control information, such that the speech decoder ( 40 ), based on that information. provides a dynamic adaptation of the size of the jitter buffer ( 20 ) using the received encoded, packetized speech signal. The invention also relates to a method of adapting the functional size of the jitter buffer of a receiver system.

公开(公告)号：SE528248C2

公开(公告)日：2006-10-03

申请号：SE0402988

申请日：2004-12-08

Applicant: ERICSSON TELEFON AB L M

Inventor： LUNDBERG JONAS ,  HANNU HANS ,  NOHLGREN ANDERS

IPC: G10L13/00 ,  G10L21/04 ,  H04J3/06 ,  H04L69/40 ,  H04M7/00 ,  H04W4/10 ,  H04W84/08 ,  H04W88/02

公开(公告)号：SE9902630L

公开(公告)日：2001-01-09

申请号：SE9902630

申请日：1999-07-08

Applicant: ERICSSON TELEFON AB L M

Inventor： NOHLGREN ANDERS ,  SUNDQVIST JIM

IPC: H04L29/08 ,  H04B14/04 ,  H04J3/06 ,  H04L7/00 ,  H04L12/64 ,  H04Q11/04

Abstract: The invention is concerned with a method and an apparatus, wherein information data is sent between at least two transceivers in a telecommunication system. The information data is transmitted from the sender of a transceiver to the receiver of one or more other transceivers in form of digital signals having a given sampling frequency. The signals are played out by the receiver in a controlled way. The invention is mainly characterized by estimation of the sender's sampling rate at the sending side of a transceiver, transmitting the estimation to the receiving side of an another transceiver, and controlling the playout of the information data at the receiving side by means of the sampling rate estimated at the sending side to avoid delays and/or interrupts in the presentation. The invention is especially suitable in connection with packet based networks wherein the information data is sent between the transceivers in the telecommunication system in form of packet data frames, such as audio frames.

公开(公告)号：SE9902630D0

公开(公告)日：1999-07-08

申请号：SE9902630

申请日：1999-07-08

Applicant: ERICSSON TELEFON AB L M

Inventor： NOHLGREN ANDERS ,  SUNDQVIST JIM

IPC: H04L29/08 ,  H04B14/04 ,  H04J3/06 ,  H04L7/00 ,  H04L12/64 ,  H04Q11/04 ,  H04L

Abstract: The invention is concerned with a method and an apparatus, wherein information data is sent between at least two transceivers in a telecommunication system. The information data is transmitted from the sender of a transceiver to the receiver of one or more other transceivers in form of digital signals having a given sampling frequency. The signals are played out by the receiver in a controlled way. The invention is mainly characterized by estimation of the sender's sampling rate at the sending side of a transceiver, transmitting the estimation to the receiving side of an another transceiver, and controlling the playout of the information data at the receiving side by means of the sampling rate estimated at the sending side to avoid delays and/or interrupts in the presentation. The invention is especially suitable in connection with packet based networks wherein the information data is sent between the transceivers in the telecommunication system in form of packet data frames, such as audio frames.

公开(公告)号：AT414315T

公开(公告)日：2008-11-15

申请号：AT01932448

申请日：2001-05-10

Applicant: ERICSSON TELEFON AB L M

Inventor： SVEDBERG JONAS ,  SUNDQVIST JIM ,  UVLIDEN ANDERS ,  NOHLGREN ANDERS ,  WESTERLUND MAGNUS

IPC: G10L19/00 ,  H03M7/36 ,  H04L1/00

Abstract: An improved forward error correction (FEC) technique for coding speech data provides an encoder module which primary-encodes an input speech signal using a primary synthesis model to produce primary-encoded data, and redundant-encodes the input speech signal using a redundant synthesis model to produce redundant-encoded data. A packetizer combines the primary-encoded data and the redundant-encoded data into a series of packets and transmits the packets over a packet-based network, such as an Internet Protocol (IP) network. A decoding module primary-decodes the packets using the primary synthesis model, and redundant-decodes the packets using the redundant synthesis model. The technique provides interaction between the primary synthesis model and the redundant synthesis model during and after decoding to improve the quality of a synthesized output speech signal. Such "interaction," for instance, may take the form of updating states in one model using the other model.

公开(公告)号：DE60129327T2

公开(公告)日：2008-03-20

申请号：DE60129327

申请日：2001-05-22

Applicant: ERICSSON TELEFON AB L M

Inventor： SUNDQVIST JIM ,  LENNESTAL HAKAN ,  NOHLGREN ANDERS ,  LINDQVIST MORGAN ,  WANG WEI

IPC: H04L12/64

Abstract: The present invention relates to a receiver system in a communication system supporting packet-based communicatior (e.g., an IP-network), including a receiver, speech decoder ( 40 ) and a jitter buffer ( 20 ) for handling delay variations in the reception of a speech signal consisting of packets containing frames with encoded speech. A jitter buffer controller ( 50 ) is provided for keeping information about the functional size of the jitter buffer ( 20 ) and for providing the speech decoder ( 40 ) with control information, such that the speech decoder ( 40 ), based on that information. provides a dynamic adaptation of the size of the jitter buffer ( 20 ) using the received encoded, packetized speech signal. The invention also relates to a method of adapting the functional size of the jitter buffer of a receiver system.

公开(公告)号：DE10296790T5

公开(公告)日：2004-04-22

申请号：DE10296790

申请日：2002-05-10

Applicant: ERICSSON TELEFON AB L M

Inventor： LUNDBERG JONAS ,  NOHLGREN ANDERS

IPC: G06F13/00 ,  G06F17/30

Abstract: A method for presentation of media objects of a multimedia presentation document (104) in a device (105) for presentation of such documents. The media objects are stored in storage means (102), which are remotely located from said device (105) and connected to a network infrastructure (103). The method comprises the steps of: providing said multimedia presentation document with metadata relating to properties of the media objects; processing said metadata in the presentation device (105); determining when to fetch each of said media objects from the storage means (102) to the presentation device (105) via the network infrastructure (103) based on said processing; fetching the media objects from the storage means (102) at the determined moments and presenting the fetched media objects in the presentation device (105). The invention also relates to a multimedia presentation system, a multimedia presentation document, a computer program product and a multimedia presentation device.

公开(公告)号：SE0101677L

公开(公告)日：2003-01-10

申请号：SE0101677

申请日：2001-05-11

Applicant: ERICSSON TELEFON AB L M

Inventor： LUNDBERG JONAS ,  NOHLGREN ANDERS

IPC: G06F13/00 ,  G06F17/30

Abstract: A method for presentation of media objects of a multimedia presentation document (104) in a device (105) for presentation of such documents. The media objects are stored in storage means (102), which are remotely located from said device (105) and connected to a network infrastructure (103). The method comprises the steps of: providing said multimedia presentation document with metadata relating to properties of the media objects; processing said metadata in the presentation device (105); determining when to fetch each of said media objects from the storage means (102) to the presentation device (105) via the network infrastructure (103) based on said processing; fetching the media objects from the storage means (102) at the determined moments and presenting the fetched media objects in the presentation device (105). The invention also relates to a multimedia presentation system, a multimedia presentation document, a computer program product and a multimedia presentation device.