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    • 21. 发明授权
    • Partial spectral loss concealment in transform codecs
    • 变换编解码器中部分频谱损耗隐藏
    • US07356748B2
    • 2008-04-08
    • US11011780
    • 2004-12-15
    • Anisse Taleb
    • Anisse Taleb
    • H04L1/00
    • H04L63/1441G06F21/56G06F2221/03H04B1/667H04L51/12H04L63/1483H04L63/20H04N19/48H04N19/895
    • 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.
    • 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。
    • 22. 发明授权
    • Energy conservative multi-channel audio coding
    • 节能多声道音频编码
    • US09330671B2
    • 2016-05-03
    • US13122880
    • 2009-09-25
    • Erik NorvellMartin SehlstedtAnisse Taleb
    • Erik NorvellMartin SehlstedtAnisse Taleb
    • G10L19/008
    • G10L19/008
    • 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.
    • 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。
    • 23. 发明授权
    • Low-complexity code excited linear prediction encoding
    • 低复杂度码激励线性预测编码
    • US08000967B2
    • 2011-08-16
    • US11074928
    • 2005-03-09
    • Anisse Taleb
    • Anisse Taleb
    • G10L13/00
    • G10L19/10
    • 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.
    • 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。
    • 24. 发明申请
    • Low-Complexity Spectral Analysis/Synthesis Using Selectable Time Resolution
    • 使用可选择的时间分辨率的低复杂度光谱分析/综合
    • US20100250265A1
    • 2010-09-30
    • US12675461
    • 2008-08-25
    • Anisse Taleb
    • Anisse Taleb
    • G10L21/00
    • G10L19/02G10L19/022
    • 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.
    • 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。
    • 25. 发明授权
    • Constrained filter encoding of polyphonic signals
    • 复音信号的约束滤波器编码
    • US07725324B2
    • 2010-05-25
    • US11011764
    • 2004-12-15
    • Stefan BruhnIngemar JohanssonAnisse TalebPatrik Sandgren
    • Stefan BruhnIngemar JohanssonAnisse TalebPatrik Sandgren
    • G10L19/00G10L21/04G10L19/02H04R5/00
    • G10L19/008
    • 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.
    • 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。
    • 26. 发明申请
    • Constrained filter encoding of polyphonic signals
    • 复音信号的约束滤波器编码
    • US20050160126A1
    • 2005-07-21
    • US11011764
    • 2004-12-15
    • Stefan BruhnIngemar JohanssonAnisse TalebPatrik Sandgren
    • Stefan BruhnIngemar JohanssonAnisse TalebPatrik Sandgren
    • G06F17/10
    • G10L19/008
    • 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.
    • 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。
    • 28. 发明授权
    • Low-complexity spectral analysis/synthesis using selectable time resolution
    • 使用可选择的时间分辨率的低复杂度光谱分析/综合
    • US08392202B2
    • 2013-03-05
    • US12675461
    • 2008-08-25
    • Anisse Taleb
    • Anisse Taleb
    • G10L21/00
    • G10L19/02G10L19/022
    • 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.
    • 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。