LOW-COMPLEXITY SPECTRAL ANALYSIS/SYNTHESIS USING SELECTABLE TIME RESOLUTION
    21.
    发明申请
    LOW-COMPLEXITY SPECTRAL ANALYSIS/SYNTHESIS USING SELECTABLE TIME RESOLUTION 审中-公开
    低复杂度谱分析/综合利用可选时间分辨率

    公开(公告)号:WO2009029032A2

    公开(公告)日:2009-03-05

    申请号:PCT/SE2008050959

    申请日:2008-08-25

    Inventor: TALEB ANISSE

    CPC classification number: G10L19/02 G10L19/022

    Abstract: The signal processing is based on the c oncept of using a time-domain aliased (12, TDA) frame as a basis for time segmen tation (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-frequenc y transform can thus be changed by simply adapting the time segmentation to ob tain 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别名(12,TDA)帧作为时间分段(14)和频谱分析(16)的基础的基础上,基于时域别名 并基于所得到的时间段执行频谱分析。 总体“分段”时频变换的时间分辨率因此可以通过简单地调整时间分段以基于应用哪种频谱分析来获得适当数量的时间段来改变。 针对所有片段获得的整个频谱系数集提供了原始信号帧的可选择的时间 - 频率平铺。

    METHOD AND ARRANGEMENT FOR A DECODER FOR MULTI-CHANNEL SURROUND SOUND
    22.
    发明申请
    METHOD AND ARRANGEMENT FOR A DECODER FOR MULTI-CHANNEL SURROUND SOUND 审中-公开
    用于多通道环绕声解码器的方法和装配

    公开(公告)号:WO2007111568A2

    公开(公告)日:2007-10-04

    申请号:PCT/SE2007050194

    申请日:2007-03-28

    Inventor: TALEB ANISSE

    CPC classification number: H04S3/02 G10L19/008 H04S2420/03

    Abstract: The basic concept of the present invention is to extrapolate a partially known spatial covariance matrix of a multi-channel signal in the parameter domain. The extrapolated covariance matrix is used with the downcoded downmix signal in order to efficiently generate an estimate of a linear combination of the multi-channel signals.

    Abstract translation: 本发明的基本概念是推断参数域中的多信道信号的部分已知的空间协方差矩阵。 外推协方差矩阵与降频缩减信号一起使用,以有效地产生多信道信号的线性组合的估计。

    IMPROVED FILTER SMOOTHING IN MULTI-CHANNEL AUDIO ENCODING AND/OR DECODING
    24.
    发明申请
    IMPROVED FILTER SMOOTHING IN MULTI-CHANNEL AUDIO ENCODING AND/OR DECODING 审中-公开
    在多声道音频编码和/或解码中改进滤波器平滑度

    公开(公告)号:WO2006091150B1

    公开(公告)日:2006-12-14

    申请号:PCT/SE2006000234

    申请日:2006-02-22

    CPC classification number: G10L19/008 G10L19/24

    Abstract: A first signal representation of one or more of the multiple channels is encoded (Sl) 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: 在第一编码过程中编码(S1)多个信道中的一个或多个信道的第一信号表示,并且在第二基于滤波器的编码过程中编码(S2)多个信道中的一个或多个信道的第二信号表示 。 滤波器平滑可用于减少编码伪像的影响。 然而,传统的滤波器平滑通常导致相当大的性能降低,因此不被广泛使用。 已经认识到,编码伪像被认为比立体宽度的临时减少更令人讨厌,并且当编码滤波器提供目标信号的差估计时它们特别令人讨厌; 估计越差,更令人不安的文物。 因此,在第二编码过程或相应的解码过程中引入信号自适应滤波器平滑(S3)作为用于解决现有技术的问题的新的一般概念。

    Método y dispositivo para la descodificación espectral perceptual de una señal de audio, que incluyen el llenado de huecos espectrales

    公开(公告)号:ES2704286T3

    公开(公告)日:2019-03-15

    申请号:ES08828426

    申请日:2008-08-26

    Abstract: Método para la descodificación espectral perceptual de una señal de audio, que comprende las etapas de: descodificar (210) coeficientes espectrales recuperados de un flujo binario para obtener coeficientes espectrales descodificados de un conjunto inicial de coeficientes espectrales; llenar espectralmente (212) dicho conjunto inicial de coeficientes espectrales obteniendo un conjunto de coeficientes espectrales reconstruidos; comprendiendo dicho llenado espectral (212) un llenado (214), con ruido, de huecos espectrales mediante la fijación de coeficientes espectrales de dicho conjunto inicial de coeficientes espectrales que no se reciben en dicho flujo binario, de manera que sean iguales a elementos obtenidos a partir de dichos coeficientes espectrales descodificados; y convertir (216) dicho conjunto de coeficientes espectrales reconstruidos de un dominio de frecuencia en una señal de audio en un dominio en el tiempo, caracterizado por que dicho llenado (214) con ruido comprende, a su vez, la creación (262) de un libro de códigos espectral concatenando los coeficientes espectrales perceptualmente relevantes de dichos coeficientes espectrales descodificados, con lo cual dicho llenado (214), con ruido, de huecos espectrales comprende la fijación de coeficientes espectrales en dicho conjunto inicial de coeficientes espectrales de manera que sean iguales a elementos seleccionados (266) de dicho libro de códigos espectral de acuerdo con por lo menos un criterio; uno del por lo menos un criterio es seleccionar (266) elementos de dicho libro de códigos espectral en un orden de índices comenzando desde el extremo de baja frecuencia, en donde se asignan índices i a los coeficientes espectrales y se asignan índices j a los elementos del libro de códigos espectral, en donde los huecos espectrales se llenan a ciegas incrementando el índice j en la misma medida que el índice i, y mediante un uso cíclico del libro de códigos espectral en caso de que haya más huecos espectrales que elementos en el libro de códigos espectral.

    METHOD AND DEVICE FOR NOISE FILLING

    公开(公告)号:CA2698031C

    公开(公告)日:2016-10-18

    申请号:CA2698031

    申请日:2008-08-26

    Abstract: A method for perceptual spectral decoding comprises decoding of spectral coefficients recovered from a binary flux into decoded spectral coefficients of an initial set of spectral coefficients. The initial set of spectral coefficients are spectrum filled. The spectrum filling comprises noise filling of spectral holes by setting spectral coefficients in the initial set of spectral coefficients not being decoded from the binary flux equal to elements derived from the decoded spectral coefficients. The set of reconstructed spectral coefficients of a frequency domain formed by the spectrum filling is converted into an audio signal of a time domain. A perceptual spectral decoder comprises a noise filler, operating according to the method for perceptual spectral decoding.

    28.
    发明专利
    未知

    公开(公告)号:AT538604T

    公开(公告)日:2012-01-15

    申请号:AT07716149

    申请日:2007-03-28

    Inventor: TALEB ANISSE

    Abstract: The basic concept of the present invention is to extrapolate a partially known spatial covariance matrix of a multi-channel signal in the parameter domain. The extrapolated covariance matrix is used with the downcoded downmix signal in order to efficiently generate an estimate of a linear combination of the multi-channel signals.

    29.
    发明专利
    未知

    公开(公告)号:AT521143T

    公开(公告)日:2011-09-15

    申请号:AT05822014

    申请日:2005-12-22

    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.

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