公开(公告)号：WO2006091151B1

公开(公告)日：2006-12-14

申请号：PCT/SE2006000235

申请日：2006-02-22

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE ,  ANDERSSON STEFAN

Inventor： TALEB ANISSE ,  ANDERSSON STEFAN

IPC: G10L19/00 ,  H04B1/66

CPC classification number: G10L19/008 ,  G10L19/24

Abstract: The invention provides an efficient technique for encoding a multi-channel audio signal. The invention relies on the principle of encoding (Sl) a signal representation of one or more of the multiple channels in a first encoding process, and encoding another signal representation of one or more channels in a second, filter-based encoding process. A basic idea according to the invention is to select (S2), for the second encoding process, a combination of i) frame division configuration of an overall encoding frame into a set of sub-frames, and ii) filter length for each sub-frame, according to a predetermined criterion. The second signal representation is then encoded (S3) in each sub-frame of the overall encoding frame according to the selected combination. The possibility to select frame division configuration and at the same time adjust the filter length for each sub-frame provides added degrees of freedom, and generally results in improved performance.

Abstract translation: 本发明提供了一种用于对多声道音频信号进行编码的有效技术。 本发明依赖于在第一编码过程中编码（S1）一个或多个多个信道的信号表示的原理，以及在第二个基于过滤器的编码过程中编码一个或多个信道的另一个信号表示。 根据本发明的基本思想是，对于第二编码处理，选择（S2）i）将整个编码帧的帧分配配置成一组子帧的组合，以及ii）每个子帧的滤波器长度， 帧，根据预定标准。 然后根据所选择的组合，在整个编码帧的每个子帧中对第二信号表示进行编码（S3）。 选择帧分配配置并同时调整每个子帧的滤波器长度的可​​能性提供了附加的自由度，并且通常导致改进的性能。

公开(公告)号：WO2007035148A3

公开(公告)日：2007-05-10

申请号：PCT/SE2006001043

申请日：2006-09-12

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE

Inventor： TALEB ANISSE

IPC: H03M7/30

CPC classification number: H03M7/3082

Abstract: A vector quantizer includes a lattice quantizer (10) approximating a vector x by a lattice vector belonging to a lattice ? 0 . A lattice vector decomposer (14) connected to the lattice quantizer successively decomposes the lattice vector into a sequence of quotient vectors y, and a sequence of remainder vectors r i on successive lattices ? I-1 by lattice division with a corresponding predetermined sequence of integers p

Abstract translation: 矢量量化器包括通过属于格子φ0的晶格矢量近似矢量x的晶格量化器（10）。 连接到晶格量化器的晶格矢量分解器（14）将晶格矢量依次分解成一个商矢量y的序列，并且在连续晶格上的剩余矢量序列ΠI-1 通过与相应的预定的整数序列p进行点划分

公开(公告)号：WO2009029032A2

公开(公告)日：2009-03-05

申请号：PCT/SE2008050959

申请日：2008-08-25

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE

Inventor： TALEB ANISSE

IPC: G10L19/02

CPC classification number: G10L19/02 ,  G10L19/022

Abstract: The signal processing is based on the c oncept of using a time-domain aliased (12, TDA) frame as a basis for time segmen tation (14) and spectral analysis (16), performing segmentation in time based on the time-domain aliased frame and performing spectral analysis based on the resulting time segments. The time resolution of the overall "segmented" time-to-frequenc y transform can thus be changed by simply adapting the time segmentation to ob tain a suitable number of time segments based on which spectral analysis is applied. The overall set of spectral coefficients, obtained for all the segments, provides a selectable time-frequency tiling of the original signal frame.

Abstract translation: 信号处理基于使用时域别名（12，TDA）帧作为时间分段（14）和频谱分析（16）的基础的基础上，基于时域别名 并基于所得到的时间段执行频谱分析。 总体“分段”时频变换的时间分辨率因此可以通过简单地调整时间分段以基于应用哪种频谱分析来获得适当数量的时间段来改变。 针对所有片段获得的整个频谱系数集提供了原始信号帧的可选择的时间 - 频率平铺。

公开(公告)号：WO2007111568A2

公开(公告)日：2007-10-04

申请号：PCT/SE2007050194

申请日：2007-03-28

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE

Inventor： TALEB ANISSE

CPC classification number: H04S3/02 ,  G10L19/008 ,  H04S2420/03

Abstract: The basic concept of the present invention is to extrapolate a partially known spatial covariance matrix of a multi-channel signal in the parameter domain. The extrapolated covariance matrix is used with the downcoded downmix signal in order to efficiently generate an estimate of a linear combination of the multi-channel signals.

Abstract translation: 本发明的基本概念是推断参数域中的多信道信号的部分已知的空间协方差矩阵。 外推协方差矩阵与降频缩减信号一起使用，以有效地产生多信道信号的线性组合的估计。

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：WO2006091150B1

公开(公告)日：2006-12-14

申请号：PCT/SE2006000234

申请日：2006-02-22

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE ,  ANDERSSON STEFAN

Inventor： TALEB ANISSE ,  ANDERSSON STEFAN

IPC: G10L19/00 ,  H04B1/66

CPC classification number: G10L19/008 ,  G10L19/24

Abstract: A first signal representation of one or more of the multiple channels is encoded (Sl) in a first encoding process, and a second signal representation of one or more of the multiple channels is encoded (S2) in a second, filter-based encoding process. Filter smoothing can be used to reduce the effects of coding artifacts. However, conventional filter smoothing generally leads to a rather large performance reduction and is therefore not widely used. It has been recognized that coding artifacts are perceived as more annoying than temporary reduction in stereo width, and that they are especially annoying when the coding filter provides a poor estimate of the target signal; the poorer the estimate, the more disturbing artifacts. Therefore, signal-adaptive filter smoothing (S3) is introduced in the second encoding process or a corresponding decoding process as a new general concept for solving the problems of the prior art.

Abstract translation: 在第一编码过程中编码（S1）多个信道中的一个或多个信道的第一信号表示，并且在第二基于滤波器的编码过程中编码（S2）多个信道中的一个或多个信道的第二信号表示 。 滤波器平滑可用于减少编码伪像的影响。 然而，传统的滤波器平滑通常导致相当大的性能降低，因此不被广泛使用。 已经认识到，编码伪像被认为比立体宽度的临时减少更令人讨厌，并且当编码滤波器提供目标信号的差估计时它们特别令人讨厌; 估计越差，更令人不安的文物。 因此，在第二编码过程或相应的解码过程中引入信号自适应滤波器平滑（S3）作为用于解决现有技术的问题的新的一般概念。

公开(公告)号：WO2007111560A3

公开(公告)日：2007-12-13

申请号：PCT/SE2007000299

申请日：2007-03-28

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE ,  POBLOTH HARALD ,  KARLSSON ERLENDUR ,  NORVELL ERIK

Inventor： TALEB ANISSE ,  POBLOTH HARALD ,  KARLSSON ERLENDUR ,  NORVELL ERIK

IPC: H03H21/00 ,  H04R3/12 ,  H04S3/00

CPC classification number: H04S3/00 ,  H03H2017/0072 ,  H04S2400/01 ,  H04S2420/01 ,  H04S2420/03

Abstract: In signal processing using digital filtering the representation of a filter is adapted depending on the filter characteristics. If e.g. a digital filter is represented by filter coefficients for transform bands Nos. 0, 1,..., K in the frequency domain, a reduced digital filter having coefficients for combined transform bands, i.e. subsets of the transformed bands, Nos. 0, 1,..., L, is formed and only these coefficients are stored. When the actual filtering in the digital filter is to be performed, an actual digital filter is obtained by expanding the coefficients of the reduced digital filter according to a mapping table and then used instead of the original digital filter.

Abstract translation: 在使用数字滤波的信号处理中，根据滤波器特性来调整滤波器的表示。 如果例如 数字滤波器由频域中的变换频带0,1，...，K的滤波器系数表示，具有用于组合变换频带的系数的缩减数字滤波器，即变换频带的子集0,1 ，...，L，并且只有这些系数被存储。 当要进行数字滤波器中的实际滤波时，通过根据映射表扩展减小的数字滤波器的系数，然后使用替代原来的数字滤波器来获得实际的数字滤波器。

公开(公告)号：WO2007102782A3

公开(公告)日：2007-11-08

申请号：PCT/SE2007050132

申请日：2007-03-07

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE

Inventor： TALEB ANISSE

IPC: G10L19/14 ,  G10L19/04

CPC classification number: G10L19/06 ,  G10L19/04 ,  G10L19/24

Abstract: A method for audio coding and decoding comprises primary encoding (12) of a present audio signal sample into an encoded representation (T(n)), and non-causal encoding (13) of a first previous audio signal sample into an encoded enhancement representation (ET(n-N+)). The method further comprises providing of the encoded representations to an end user. At the end user, the method comprises primary decoding (52) of the encoded representation (T*(n)) into a present received audio signal sample, and non- causal decoding (53) of the encoded enhancement representation (ET*(n-N+)) into an enhancement first previous received audio signal sample. The method further comprises improving of a first previous received audio signal sample, corresponding to the first previous audio signal sample, based on the enhancement first previous received audio signal sample. Devices and systems for audio coding and decoding are also presented.

Abstract translation: 一种用于音频编码和解码的方法包括：将当前音频信号样本初级编码（12）为编码表示（T（n）），并将第一先前音频信号样本的非因果编码（13）编码为编码增强表示 （ET（N-N +））。 该方法还包括将编码表示提供给最终用户。 在最终用户处，该方法包括将编码表示（T *（n））初级解码（52）为当前接收音频信号采样，并且对编码增强表示（ET *（n）的非因果解码（53） -N +））转换成增强的第一先前接收音频信号样本。 该方法进一步包括基于增强的第一先前接收音频信号样本来改善对应于第一先前音频信号样本的第一先前接收音频信号样本。 还介绍了用于音频编码和解码的设备和系统。

公开(公告)号：WO2007078254A2

公开(公告)日：2007-07-12

申请号：PCT/SE2007000006

申请日：2007-01-05

Applicant: ERICSSON TELEFON AB L M ,  TALEB ANISSE ,  KARLSSON ERLENDUR

Inventor： TALEB ANISSE ,  KARLSSON ERLENDUR

IPC: H04S3/00 ,  G10L19/00 ,  G10L19/008

CPC classification number: G10L19/008 ,  H04R2420/07 ,  H04S3/004 ,  H04S3/008 ,  H04S2420/03

Abstract: A parametric multi-channel surround audio bitstream is received in a multi-channel decoder (13). The received spatial parameters are in combining unit (37) transformed into a new set of spatial parameters that are used in order to obtain a decoding of the multi-channel surround sound that is not a simple equivalent of the original input multi-channel surround signal but e.g. may be personalized by making the transformation based on a representation of user head related filters obtained from a unit (43). Such personalized spatial parameters may also be obtained by combining the received spatial parameters and a representation of the user head related filters with a set of additional rendering parameters that for example are interactively determined by the user and thus are time dependent.

Abstract translation: 参数多声道环绕音频比特流被接收在多声道解码器（13）中。 所接收的空间参数在被转换成新的一组空间参数的组合单元（37）中，这些空间参数被用于获得不是原始输入多声道环绕信号的简单等价物的多声道环绕声的解码 但例如 可以通过基于从单元（43）获得的与用户头相关的过滤器的表示进行变换来进行个性化。 这样的个性化空间参数也可以通过将接收到的空间参数和用户头部相关过滤器的表示与例如由用户交互地确定并且因此是时间相关的一组附加渲染参数组合来获得。

公开(公告)号：EP2186090A4

公开(公告)日：2013-12-25

申请号：EP08828880

申请日：2008-08-25

Applicant: ERICSSON TELEFON AB L M

Inventor： TALEB ANISSE ,  ULLBERG GUSTAF

IPC: G10L19/02

CPC classification number: G10L19/025 ,  G10L19/0212

Abstract: A transient detector (100) analyzes (110) a given frame n of the input audio signal to determine, based on audio signal characteristics of the given frame n, a transient hangover indicator for a following frame n+1, and signals (120) the determined transient hangover indicator to an associated audio encoder (10) to enable proper encoding of the following frame n+1.