公开(公告)号：US07822617B2

公开(公告)日：2010-10-26

申请号：US11358726

申请日：2006-02-22

Applicant: Anisse Taleb ,  Stefan Andersson

Inventor： Anisse Taleb ,  Stefan Andersson

IPC: G10L19/00

CPC classification number: G10L19/022 ,  G10L19/002 ,  G10L19/008 ,  G10L19/24 ,  G10L19/26

Abstract: The invention provides an efficient technique for encoding a multi-channel audio signal. The invention relies on the principle of encoding (S1) 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）多个信道中的一个或多个的信号表示的原理，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选择（S2）对于第二编码处理，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：US20080262850A1

公开(公告)日：2008-10-23

申请号：US11816996

申请日：2005-12-22

Applicant: Anisse Taleb ,  Stefan Andersson

Inventor： Anisse Taleb ,  Stefan Andersson

IPC: G10L19/00

CPC classification number: G10L19/008 ,  G10L19/24

Abstract: The invention provides a highly efficient technique for encoding a multi-channel audio signal. The invention relies on the basic principle of encoding a first signal representation of one or more of the multiple channels in a first encoder (130) and encoding a second signal representation of one or more of the multiple channels in a second, multi-stage, encoder (140). This procedure is significantly enhanced by providing a controller (150) for adaptively allocating a number of encoding bits among the different encoding stages of the second, multi-stage, encoder (140) in dependence on multi-channel audio signal characteristics.

Abstract translation: 本发明提供了一种用于编码多声道音频信号的高效技术。 本发明依赖于在第一编码器（130）中编码多个信道中的一个或多个信道的第一信号表示的基本原理，并且在第二多级信道中编码多个信道中的一个或多个信道的第二信号表示， 编码器（140）。 通过提供一种用于根据多声道音频信号特性在第二，多级编码器（140）的不同编码级之间自适应地分配多个编码位的控制器（150）来显着增强该过程。

公开(公告)号：US20060246868A1

公开(公告)日：2006-11-02

申请号：US11358720

申请日：2006-02-22

Applicant: Anisse Taleb ,  Stefan Andersson

Inventor： Anisse Taleb ,  Stefan Andersson

IPC: H04B1/10

CPC classification number: G10L19/022 ,  G10L19/002 ,  G10L19/008 ,  G10L19/24 ,  G10L19/26

Abstract: A first signal representation of one or more of the multiple channels is encoded (S1) 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: 在第一编码处理中对多个信道中的一个或多个信道的第一信号表示进行编码（S 1），并且第二编码（S 2）中的一个或多个多信道的第二信号表示，基于过滤器 编码过程。 滤波平滑可用于减少编码伪像的影响。 然而，常规滤波器平滑通常导致相当大的性能降低，并且因此不被广泛使用。 已经认识到，编码伪像被认为比立体声宽度的暂时减少更烦人，并且当编码滤波器提供对目标信号的不良估计时，它们特别烦人; 估计越穷越好的文物。 因此，在第二编码处理或对应的解码处理中引入信号自适应滤波平滑（S 3）作为解决现有技术问题的新的一般概念。

公开(公告)号：US09626973B2

公开(公告)日：2017-04-18

申请号：US11816996

申请日：2005-12-22

Applicant: Anisse Taleb ,  Stefan Andersson

Inventor： Anisse Taleb ,  Stefan Andersson

IPC: G10L19/008 ,  G10L19/24

CPC classification number: G10L19/008 ,  G10L19/24

Abstract: The invention provides a highly efficient technique for encoding a multi-channel audio signal. The invention relies on the basic principle of encoding a first signal representation of one or more of the multiple channels in a first encoder and encoding a second signal representation of one or more of the multiple channels in a second, multi-stage, encoder. This procedure is significantly enhanced by providing a controller for adaptively allocating a number of encoding bits among the different encoding stages of the second, multi-stage, encoder in dependence on multi-channel audio signal characteristics.

公开(公告)号：US07945055B2

公开(公告)日：2011-05-17

申请号：US11358720

申请日：2006-02-22

Applicant: Anisse Taleb ,  Stefan Andersson

Inventor： Anisse Taleb ,  Stefan Andersson

IPC: H04R5/00

CPC classification number: G10L19/022 ,  G10L19/002 ,  G10L19/008 ,  G10L19/24 ,  G10L19/26

Abstract: A first signal representation of one or more of the multiple channels is encoded in a first encoding process, and a second signal representation of one or more of the multiple channels is encoded 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 is introduced in the second encoding process or a corresponding decoding process.

Abstract translation: 在第一编码过程中编码多个信道中的一个或多个的第一信号表示，并且在第二个基于过滤器的编码过程中对多个信道中的一个或多个的第二信号表示进行编码。 滤波平滑可用于减少编码伪像的影响。 然而，常规滤波器平滑通常导致相当大的性能降低，并且因此不被广泛使用。 已经认识到，编码伪像被认为比立体声宽度的暂时减少更烦人，并且当编码滤波器提供对目标信号的不良估计时，它们特别烦人; 估计越穷越好的文物。 因此，在第二编码处理或对应的解码处理中引入信号自适应滤波平滑。

公开(公告)号：US20060195314A1

公开(公告)日：2006-08-31

申请号：US11358726

申请日：2006-02-22

Applicant: Anisse Taleb ,  Stefan Andersson

Inventor： Anisse Taleb ,  Stefan Andersson

IPC: G10L19/00

CPC classification number: G10L19/022 ,  G10L19/002 ,  G10L19/008 ,  G10L19/24 ,  G10L19/26

Abstract: The invention provides an efficient technique for encoding a multi-channel audio signal. The invention relies on the principle of encoding (S1) 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（S 1）一个或多个多个信道的信号表示的原理，并且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（S 2）i）整体编码帧的帧分配配置到一组子帧的组合，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（S 3）。 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：US09495971B2

公开(公告)日：2016-11-15

申请号：US12673862

申请日：2008-08-25

Applicant: Gustaf Ullberg ,  Anisse Taleb

Inventor： Gustaf Ullberg ,  Anisse Taleb

IPC: G10L19/025 ,  G10L19/02 ,  G10L25/21

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.

Abstract translation: 瞬态检测器（100）分析（110）输入音频信号的给定帧n，以基于给定帧n的音频信号特性来确定用于后续帧n + 1的瞬时宿醉指示符，以及信号（120） 将所确定的瞬时停播指示符发送到相关联的音频编码器（10），以便能够对后续帧n + 1进行适当的编码。

公开(公告)号：US20110292924A1

公开(公告)日：2011-12-01

申请号：US13195315

申请日：2011-08-01

Applicant: Anisse Taleb ,  Stefan Bruhn ,  Moo Young Kim ,  Ingemar Johansson ,  Stefan Häkansson ,  Klas Fredrik Janis Jansson

Inventor： Anisse Taleb ,  Stefan Bruhn ,  Moo Young Kim ,  Ingemar Johansson ,  Stefan Häkansson ,  Klas Fredrik Janis Jansson

IPC: H04W4/00 ,  H04W84/02

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: 本文描述了可以：（1）在第一异构链路（例如，无线链路）上接收分组的网络处理节点（例如，MGW，MRFP）和方法; （2）基于关于第二异构链路（例如，“因特网”链路）的已知特性来操纵接收的分组; 和（3）在第二异构链路（例如，“因特网”链路）上发送所操纵的分组。 例如，网络处理节点可以通过添加冗余，去除冗余，帧聚合（重新分组），恢复丢失的分组和/或重新发送分组来操纵所接收的分组。

公开(公告)号：US20090076830A1

公开(公告)日：2009-03-19

申请号：US12281953

申请日：2007-03-07

Applicant: Anisse Taleb

Inventor： Anisse Taleb

IPC: G10L19/00

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

Abstract: A method for audio coding and decoding comprises primary encoding of a present audio signal sample into an encoded representation (T(n)), and non-causal encoding 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 of the encoded representation (T*(n)) into a present received audio signal sample, and non-causal decoding 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为编码表示（T（n）），以及将第一先前音频信号样本的非因果编码成编码增强表示（ET（n-N + ））。 该方法还包括向最终用户提供经编码的表示。 在最终用户中，该方法包括将编码表示（T *（n））的主要解码成当前接收的音频信号样本，并将编码的增强表示（ET *（n-N +））的非因果解码转换为 增强先前接收的音频信号样本。 该方法还包括基于增强的先前接收的音频信号样本来改进对应于第一先前音频信号采样的第一先前接收音频信号采样。 还提供了用于音频编码和解码的设备和系统。

公开(公告)号：US08340450B2

公开(公告)日：2012-12-25

申请号：US11991539

申请日：2006-09-12

Applicant: Anisse Taleb

Inventor： Anisse Taleb

IPC: G06K9/00 ,  H03M7/00 ,  H03M7/38

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 Λ0. 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 ri on successive lattices ΛI−1 by lattice division with a corresponding predetermined sequence of integers pi≧2, where i=1 . . . k and k is a positive integer representing the number of elements in each sequence.

Abstract translation: 矢量量化器包括通过属于晶格Λ0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格子矢量分解器（14）将格子矢量依次分解为商矢量y的序列，并且通过与相应的预定的整数序列ΛI= 2，其中i = 1。 。 。 k和k是表示每个序列中的元素数量的正整数。