公开(公告)号：US07356748B2

公开(公告)日：2008-04-08

申请号：US11011780

申请日：2004-12-15

Applicant: Anisse Taleb

Inventor： Anisse Taleb

IPC: H04L1/00

CPC classification number: H04L63/1441 ,  G06F21/56 ,  G06F2221/03 ,  H04B1/667 ,  H04L51/12 ,  H04L63/1483 ,  H04L63/20 ,  H04N19/48 ,  H04N19/895

Abstract: 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.

Abstract translation: 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。

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：US09330671B2

公开(公告)日：2016-05-03

申请号：US13122880

申请日：2009-09-25

Applicant: Erik Norvell ,  Martin Sehlstedt ,  Anisse Taleb

Inventor： Erik Norvell ,  Martin Sehlstedt ,  Anisse Taleb

IPC: G10L19/008

CPC classification number: G10L19/008

Abstract: 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.

Abstract translation: 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。

公开(公告)号：US08000967B2

公开(公告)日：2011-08-16

申请号：US11074928

申请日：2005-03-09

Applicant: Anisse Taleb

Inventor： Anisse Taleb

IPC: G10L13/00

CPC classification number: G10L19/10

Abstract: 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.

Abstract translation: 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。

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：US20100250265A1

公开(公告)日：2010-09-30

申请号：US12675461

申请日：2008-08-25

Applicant: Anisse Taleb

Inventor： Anisse Taleb

IPC: G10L21/00

CPC classification number: G10L19/02 ,  G10L19/022

Abstract: 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.

Abstract translation: 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。

公开(公告)号：US07725324B2

公开(公告)日：2010-05-25

申请号：US11011764

申请日：2004-12-15

Applicant: Stefan Bruhn ,  Ingemar Johansson ,  Anisse Taleb ,  Patrik Sandgren

Inventor： Stefan Bruhn ,  Ingemar Johansson ,  Anisse Taleb ,  Patrik Sandgren

IPC: G10L19/00 ,  G10L21/04 ,  G10L19/02 ,  H04R5/00

CPC classification number: G10L19/008

Abstract: Signals of different channels are combined into one mono signal. A set of adaptive filters, preferably one for each channel, is derived in a respective filter adaptation unit. When an adaptive filter is applied to the mono signal it reconstructs the signal of the respective channel under a perceptual constraint. The perceptual constraint is a gain and/or shape constraint. The gain constraint allows the preservation of the relative energy between the channels while the shape constraint allows more stability by avoiding unnecessary filtering of spectral nulls. The transmitted parameters are the mono signal, in encoded form, and the parameters of the adaptive filters, preferably also encoded. The receiver reconstructs the signal of the different channels by applying the adaptive filters and possibly some additional post-processing.

Abstract translation: 不同通道的信号被组合成一个单声道信号。 在相应的滤波器适配单元中导出一组适用于每个信道的自适应滤波器。 当自适应滤波器被应用于单声道信号时，它在感知约束下重构相应频道的信号。 感知约束是增益和/或形状约束。 增益约束允许保持通道之间的相对能量，而形状约束允许通过避免频谱零点的不必要的过滤来获得更多的稳定性。 传输的参数是以编码形式的单声道信号，并且自适应滤波器的参数优选地也被编码。 接收机通过应用自适应滤波器和可能的一些额外的后处理来重构不同信道的信号。

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：US20050160126A1

公开(公告)日：2005-07-21

申请号：US11011764

申请日：2004-12-15

Applicant: Stefan Bruhn ,  Ingemar Johansson ,  Anisse Taleb ,  Patrik Sandgren

Inventor： Stefan Bruhn ,  Ingemar Johansson ,  Anisse Taleb ,  Patrik Sandgren

IPC: G06F17/10

CPC classification number: G10L19/008

Abstract: Signals of different channels (C1-CN) are combined into one mono signal (x). A set of adaptive filters, preferably one for each channel (C1-CN), is derived in a respective filter adaptation unit (30:1-30:N). When an adaptive filter is applied to the mono signal (x) it reconstructs the signal of the respective channel (C1-CN) under a perceptual constraint. The perceptual constraint is a gain and/or shape constraint. The gain constraint allows the preservation of the relative energy between the channels (C1-CN) while the shape constraint allows more stability by avoiding unnecessary filtering of spectral nulls. The transmitted parameters are the mono signal (x), in encoded form, and the parameters (p1-pN) of the adaptive filters, preferably also encoded. The receiver reconstructs the signal of the different channels by applying the adaptive filters and possibly some additional post-processing.

Abstract translation: 不同信道（C 1 -C 3 N N）的信号被组合成一个单声道信号（x）。 在各自的滤波器适配单元（30：1-30：N）中导出一组自适应滤波器，优选地每个信道一个（C 1 -C N N N） 。 当自适应滤波器被应用于单声道信号（x）时，它在感知约束下重构相应信道的信号（C 1 -C 1 -C N N）。 感知约束是增益和/或形状约束。 增益约束允许保持通道之间的相对能量（C 1 -C 3 N N），同时形状约束允许通过避免频谱零点的不必要的过滤而获得更多的稳定性。 所传输的参数是编码形式的单声道信号（x），并且自适应滤波器的参数（p <1> N＆gt; N＆lt; N＆gt;）优选地也被编码。 接收机通过应用自适应滤波器和可能的一些额外的后处理来重构不同信道的信号。