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1. 发明授权

US09495971B2 Transient detector and method for supporting encoding of an audio signal 有权
标题翻译：用于支持音频信号编码的瞬态检测器和方法
公开(公告)号：US09495971B2
公开(公告)日：2016-11-15
申请号：US12673862
申请日：2008-08-25
申请人： Gustaf Ullberg , Anisse Taleb
发明人： Gustaf Ullberg , Anisse Taleb
IPC分类号： G10L19/025 , G10L19/02 , G10L25/21
CPC分类号： G10L19/025 , G10L19/0212
摘要： 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.
摘要翻译：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。

2. 发明申请

US20110292924A1 NETWORK PROCESSING NODE AND METHOD FOR MANIPULATING PACKETS 有权
标题翻译：网络处理节点和操作分组的方法
公开(公告)号：US20110292924A1
公开(公告)日：2011-12-01
申请号：US13195315
申请日：2011-08-01
申请人： Anisse Taleb , Stefan Bruhn , Moo Young Kim , Ingemar Johansson , Stefan Häkansson , Klas Fredrik Janis Jansson
发明人： Anisse Taleb , Stefan Bruhn , Moo Young Kim , Ingemar Johansson , Stefan Häkansson , Klas Fredrik Janis Jansson
IPC分类号： H04W4/00 , H04W84/02
CPC分类号： H04L1/0041 , H04L1/0009 , H04L1/0026 , H04L1/18 , H04L49/552 , H04L2001/0097
摘要： 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.
摘要翻译：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。

3. 发明授权

US07822617B2 Optimized fidelity and reduced signaling in multi-channel audio encoding 有权
标题翻译：多声道音频编码中优化的保真度和减少的信令
公开(公告)号：US07822617B2
公开(公告)日：2010-10-26
申请号：US11358726
申请日：2006-02-22
申请人： Anisse Taleb , Stefan Andersson
发明人： Anisse Taleb , Stefan Andersson
IPC分类号： G10L19/00
CPC分类号： G10L19/022 , G10L19/002 , G10L19/008 , G10L19/24 , G10L19/26
摘要： 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.
摘要翻译：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。

4. 发明申请

US20090076830A1 Methods and Arrangements for Audio Coding and Decoding 有权
标题翻译：音频编码和解码的方法和布置
公开(公告)号：US20090076830A1
公开(公告)日：2009-03-19
申请号：US12281953
申请日：2007-03-07
申请人： Anisse Taleb
发明人： Anisse Taleb
IPC分类号： G10L19/00
CPC分类号： G10L19/06 , G10L19/04 , G10L19/24
摘要： 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.
摘要翻译：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。

5. 发明申请

US20080262850A1 Adaptive Bit Allocation for Multi-Channel Audio Encoding 有权
标题翻译：适用于多通道音频编码的位分配
公开(公告)号：US20080262850A1
公开(公告)日：2008-10-23
申请号：US11816996
申请日：2005-12-22
申请人： Anisse Taleb , Stefan Andersson
发明人： Anisse Taleb , Stefan Andersson
IPC分类号： G10L19/00
CPC分类号： G10L19/008 , G10L19/24
摘要： 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.
摘要翻译：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。

6. 发明授权

US08340450B2 Successively refinable lattice vector quantization 有权
标题翻译：连续可优化的格子矢量量化
公开(公告)号：US08340450B2
公开(公告)日：2012-12-25
申请号：US11991539
申请日：2006-09-12
申请人： Anisse Taleb
发明人： Anisse Taleb
IPC分类号： G06K9/00 , H03M7/00 , H03M7/38
CPC分类号： H03M7/3082
摘要： 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.
摘要翻译：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。

7. 发明授权

US08266195B2 Filter adaptive frequency resolution 有权
标题翻译：过滤自适应频率分辨率
公开(公告)号：US08266195B2
公开(公告)日：2012-09-11
申请号：US12295110
申请日：2007-03-28
申请人： Anisse Taleb , Harald Pobloth , Erlendur Karlsson , Erik Norvell
发明人： Anisse Taleb , Harald Pobloth , Erlendur Karlsson , Erik Norvell
IPC分类号： G06F17/10
CPC分类号： H04S3/00 , H03H2017/0072 , H04S2400/01 , H04S2420/01 , H04S2420/03
摘要： 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.
摘要翻译：在使用数字滤波的信号处理中，滤波器的表示根据滤波器特性进行调整。数字滤波器由变换频带编号0，1，...的滤波器系数表示。。。在频域中的K，具有用于组合变换频带的系数的减小的数字滤波器，即变换频带的子集，编号0，1，...。。。，L，并且仅存储这些系数。当要执行数字滤波器中的实际滤波时，通过根据映射表扩展缩小数字滤波器的系数，然后代替原始数字滤波器来获得实际的数字滤波器。

8. 发明申请

US20100322429A1 Joint Enhancement of Multi-Channel Audio 有权
标题翻译：联合增强多声道音频
公开(公告)号：US20100322429A1
公开(公告)日：2010-12-23
申请号：US12677383
申请日：2008-04-17
申请人： Erik Norvell , Anisse Taleb
发明人： Erik Norvell , Anisse Taleb
IPC分类号： H04R5/00
CPC分类号： G10L19/24 , G10L19/008
摘要： An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.
摘要翻译：提出了一种整体编码过程和相关的解码过程。编码过程涉及对一组音频输入通道的信号表示进行操作的至少两个信号编码处理。本地合成与第一编码处理结合使用以产生本地解码的信号，包括第一编码过程的编码误差的表示。该本地解码信号作为输入应用于第二编码处理。整个编码过程从至少一个所述编码过程产生至少两个残留编码错误信号，包括至少所述第二编码处理。然后，残留误差信号在进一步的编码处理中进行复合残差编码，优选地基于残差误差信号之间的相关性。

9. 发明申请

US20060206319A1 Low-complexity code excited linear prediction encoding 有权
标题翻译：低复杂度码激励线性预测编码
公开(公告)号：US20060206319A1
公开(公告)日：2006-09-14
申请号：US11074928
申请日：2005-03-09
申请人： Anisse Taleb
发明人： Anisse Taleb
IPC分类号： G10L19/12
CPC分类号： G10L19/10
摘要： Information about excitation signals of a first signal encoded by CELP is used to derive a limited set of candidate excitation signals for a second correlated second signal. Preferably, pulse locations of the excitation signals of the first encoded signal are used for determining the set of candidate excitation signals. More preferably, the pulse locations of the set of candidate excitation signals are positioned in the vicinity of the pulse locations of the excitation signals of the first encoded signal. The first and second signals may be multi-channel signals of a common speech or audio signal. However, the first and second signals may also be identical, whereby the coding of the second signal can be utilized for re-encoding at a lower bit rate.
摘要翻译：关于由CELP编码的第一信号的激励信号的信息用于导出用于第二相关第二信号的候选激励信号的有限集合。优选地，第一编码信号的激励信号的脉冲位置用于确定候选激励信号的集合。更优选地，该组候选激励信号的脉冲位置位于第一编码信号的激励信号的脉冲位置附近。第一和第二信号可以是公共语音或音频信号的多声道信号。然而，第一和第二信号也可以是相同的，由此第二信号的编码可以用于以较低的比特率进行重新编码。

10. 发明申请

US20060093048A9 Partial Spectral Loss Concealment In Transform Codecs 有权
标题翻译：变换编解码器部分光谱损失隐藏
公开(公告)号：US20060093048A9
公开(公告)日：2006-05-04
申请号：US11011780
申请日：2004-12-15
申请人： Anisse Taleb
发明人： Anisse Taleb
IPC分类号： H04B1/66
CPC分类号： H04L63/1441 , G06F21/56 , G06F2221/03 , H04B1/667 , H04L51/12 , H04L63/1483 , H04L63/20 , H04N19/48 , H04N19/895
摘要： The invention concerns a frequency-domain error concealment technique for information that is represented, on a frame-by-frame basis, by coding coefficients. The basic idea is to conceal an erroneous coding coefficient by exploiting coding coefficient correlation in both time and frequency. The technique is applicable to any information, such as audio, video and image data, that is compressed into coding coefficients and transmitted under adverse channel conditions. The error concealment technique proposed by the invention has the clear advantage of exploiting the redundancy of the original information signal in time as well as frequency. For example, this offers the possibility to exploit redundancy between frames (inter-frame) as well as within frames (intra-frame). The use of coding coefficients from the same frame as the erroneous coding coefficient is sometimes referred to as intra-frame coefficient correlation and it is a special case of the more general frequency correlation.
摘要翻译：本发明涉及用于通过编码系数逐帧地表示的信息的频域错误隐藏技术。基本思想是通过利用时间和频率上的编码系数相关性来隐藏错误的编码系数。该技术适用于压缩成编码系数并在不利信道条件下发送的任何信息，如音频，视频和图像数据。本发明提出的错误隐藏技术具有在时间和频率上开发原始信息信号冗余的明显优点。例如，这提供了利用帧（帧间）以及帧内（帧内）之间的冗余的可能性。使用与错误编码系数相同的帧的编码系数有时被称为帧内系数相关，并且是更一般的频率相关的特殊情况。

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