专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

21. 发明申请

US20090175550A1 Successively Refinable Lattice Vector Quantization 有权
标题翻译：连续可重定格子矢量量化
公开(公告)号：US20090175550A1
公开(公告)日：2009-07-09
申请号：US11991539
申请日：2006-09-12
申请人： Anisse Taleb
发明人： Anisse Taleb
IPC分类号： G06T9/00 , G06K9/36 , H04N7/26 , H04N7/28
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 Λl−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Lambda的格子矢量近似矢量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成商向量y的序列，并且通过与对应的预定的整数序列pi的格子划分，在连续的格子Lambdal-1上的剩余矢量序列ri > = 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。

22. 发明申请

US20090172060A1 FILTER ADAPTIVE FREQUENCY RESOLUTION 有权
标题翻译：滤波器自适应频率分辨率
公开(公告)号：US20090172060A1
公开(公告)日：2009-07-02
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。如果例如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。

23. 发明授权

US07356748B2 Partial spectral loss concealment in transform codecs 有权
标题翻译：变换编解码器中部分频谱损耗隐藏
公开(公告)号：US07356748B2
公开(公告)日：2008-04-08
申请号：US11011780
申请日：2004-12-15
申请人： Anisse Taleb
发明人： Anisse Taleb
IPC分类号： H04L1/00
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。

24. 发明申请

US20060246868A1 Filter smoothing in multi-channel audio encoding and/or decoding 有权
标题翻译：在多声道音频编码和/或解码中滤波平滑
公开(公告)号：US20060246868A1
公开(公告)日：2006-11-02
申请号：US11358720
申请日：2006-02-22
申请人： Anisse Taleb , Stefan Andersson
发明人： Anisse Taleb , Stefan Andersson
IPC分类号： H04B1/10
CPC分类号： G10L19/022 , G10L19/002 , G10L19/008 , G10L19/24 , G10L19/26
摘要： 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.
摘要翻译：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。

25. 发明授权

US09626973B2 Adaptive bit allocation for multi-channel audio encoding 有权
公开(公告)号：US09626973B2
公开(公告)日：2017-04-18
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： G10L19/008 , G10L19/24
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 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.

26. 发明授权

US09330671B2 Energy conservative multi-channel audio coding 有权
标题翻译：节能多声道音频编码
公开(公告)号：US09330671B2
公开(公告)日：2016-05-03
申请号：US13122880
申请日：2009-09-25
申请人： Erik Norvell , Martin Sehlstedt , Anisse Taleb
发明人： Erik Norvell , Martin Sehlstedt , Anisse Taleb
IPC分类号： G10L19/008
CPC分类号： G10L19/008
摘要： The invention relates to the technical field of audio encoding and/or decoding technologies, and thus concerns an overall encoding procedure and associated decoding procedure. The encoding procedure involves at least two signal encoding processes (S1-S3) operating on signal representations of a set of audio input channels, as well as residual encoding (S7-S8). It also involves a dedicated process (S4-S6) to estimate and encode energies of the audio input channels. Each encoding process is associated with a corresponding decoding process. In the overall decoding procedure the decoded signals from each encoding process are preferably combined such that the output channels are close to the input channels in terms of energy and/or quality. Normally, the combination step also adapts to the possible loss of one or more signal representation in part or in whole, such that the energy and quality is optimized with the signals at hand in the decoder. In this way, the overall quality of the output channels is improved.
摘要翻译：本发明涉及音频编码和/或解码技术的技术领域，因此涉及整体编码过程和相关联的解码过程。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（S1-S3）以及残差编码（S7-S8）。它还涉及专门的过程（S4-S6）来估计和编码音频输入通道的能量。每个编码过程与相应的解码过程相关联。在整个解码过程中，优选地组合来自每个编码处理的解码信号，使得输出信道在能量和/或质量方面靠近输入信道。通常，组合步骤还适应于部分或全部的一个或多个信号表示的可能损失，使得能量和质量被解码器中的手头信号优化。这样，输出通道的整体质量得到提高。

27. 发明授权

US08000967B2 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：US08000967B2
公开(公告)日：2011-08-16
申请号：US11074928
申请日：2005-03-09
申请人： Anisse Taleb
发明人： Anisse Taleb
IPC分类号： G10L13/00
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。

28. 发明授权

US07945055B2 Filter smoothing in multi-channel audio encoding and/or decoding 有权
标题翻译：在多声道音频编码和/或解码中滤波平滑
公开(公告)号：US07945055B2
公开(公告)日：2011-05-17
申请号：US11358720
申请日：2006-02-22
申请人： Anisse Taleb , Stefan Andersson
发明人： Anisse Taleb , Stefan Andersson
IPC分类号： H04R5/00
CPC分类号： G10L19/022 , G10L19/002 , G10L19/008 , G10L19/24 , G10L19/26
摘要： 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.
摘要翻译：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。

29. 发明申请

US20100250265A1 Low-Complexity Spectral Analysis/Synthesis Using Selectable Time Resolution 有权
标题翻译：使用可选择的时间分辨率的低复杂度光谱分析/综合
公开(公告)号：US20100250265A1
公开(公告)日：2010-09-30
申请号：US12675461
申请日：2008-08-25
申请人： Anisse Taleb
发明人： Anisse Taleb
IPC分类号： G10L21/00
CPC分类号： G10L19/02 , G10L19/022
摘要： The signal processing is based on the concept of using a time-domain aliased (12, TDA) frame as a basis for time segmentation (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-frequency transform can thus be changed by simply adapting the time segmentation to obtain 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.
摘要翻译：信号处理基于使用时域混叠（12，TDA）帧作为时间分割（14）和频谱分析（16）的基础的概念，基于时域混叠帧在时间上执行分割，并且基于所得到的时间段进行光谱分析。因此，通过简单地调整时间分割以基于哪个频谱分析被应用来获得合适数量的时间段，可以改变整个“分段”时间 - 频率变换的时间分辨率。对于所有段获得的总体频谱系数集合提供了原始信号帧的可选择的时频平铺。

30. 发明申请

US20060195314A1 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：US20060195314A1
公开(公告)日：2006-08-31
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（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。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式