MULTIMODE CODING OF SPEECH-LIKE AND NON-SPEECH-LIKE SIGNALS
    1.
    发明授权
    MULTIMODE CODING OF SPEECH-LIKE AND NON-SPEECH-LIKE SIGNALS 有权
    多模态编码语音喜欢的语言不同SIGNALS

    公开(公告)号:EP2269188B1

    公开(公告)日:2014-06-11

    申请号:EP09720866.4

    申请日:2009-03-12

    摘要: The invention relates to the coding of audio signals that may include both speech-like and non-speech-like signal components. It describes methods and apparatus for code excited linear prediction (CELP) audio encoding and decoding that employ linear predictive coding (LPC) synthesis filters controlled by LPC parameters, a plurality of codebooks each having codevectors, at least one codebook providing an excitation more appropriate for non-speech-like signals and at least one codebook providing an excitation more appropriate for speech-like signals, and a plurality of gain factors, each associated with a codebook. The encoding methods and apparatus select from the codebooks codevectors and/or associated gain factors by minimizing a measure of the difference between the audio signal and a reconstruction of the audio signal derived from the codebook excitations. The decoding methods and apparatus generate a reconstructed output signal from the LPC parameters, codevectors, and gain factors.

    Audio decoder and decoding method using efficient downmixing
    3.
    发明公开
    Audio decoder and decoding method using efficient downmixing 有权
    Audiodecodierung unter Verwendung von effizientem Downmixen

    公开(公告)号:EP2360683A1

    公开(公告)日:2011-08-24

    申请号:EP11154910.1

    申请日:2011-02-17

    IPC分类号: G10L19/00 H04S3/00

    摘要: A method, an apparatus, a computer readable storage medium configured with instructions for carrying out a method, and logic encoded in one or more computer-readable tangible medium to carry out actions. The method is to decode audio data that includes N.n channels to M.m decoded audio channels, including unpacking metadata and unpacking and decoding frequency domain exponent and mantissa data; determining transform coefficients from the unpacked and decoded frequency domain exponent and mantissa data; inverse transforming the frequency domain data; and in the case M

    摘要翻译: 配置有用于执行方法的指令的方法,装置,计算机可读存储介质,以及编码在一个或多个计算机可读有形介质中以执行动作的逻辑。 该方法是将包括N.n个信道的音频数据解码为M.m个解码的音频信道,包括解包元数据和解码和解码频域指数和尾数数据; 从解压缩和解码的频域指数和尾数数据确定变换系数; 逆变换频域数据; 在M

    METHOD FOR DETERMINING UPDATED FILTER COEFFICIENTS OF AN ADAPTIVE FILTER ADAPTED BY AN LMS ALGORITHM WITH PRE-WHITENING
    4.
    发明公开
    METHOD FOR DETERMINING UPDATED FILTER COEFFICIENTS OF AN ADAPTIVE FILTER ADAPTED BY AN LMS ALGORITHM WITH PRE-WHITENING 有权
    用于使用LMS算法的厘定更新后的滤波器系数法调整自适应滤波器与VORWEISSUNG

    公开(公告)号:EP2327156A1

    公开(公告)日:2011-06-01

    申请号:EP09791826.2

    申请日:2009-08-24

    摘要: The application relates to a method for determining at least one updated filter coefficient of an adaptive filter (22) adapted by an LMS algorithm. According to the method, filter coeffi-cients of a first whitening filter (25') are determined, in particular filter coefficients of an LPC whitening filter. The first whitening filter (25') generates a filtered signal. A normaliza-tion value is determined based on one or more computed values obtained in the course of determining the filter coefficients of the first whitening filter (25'). The normalization value is associated with the energy of the filtered signal. At least one updated filter coefficient of the adaptive filter (22) is determined in dependency on the filtered signal and the normaliza-tion value. Preferably, updated filter coefficients for all filter coefficients of the adaptive filter (22) are determined.

    METHOD FOR APPARATUS FOR AUDIO MATRIX DECODING
    5.
    发明授权
    METHOD FOR APPARATUS FOR AUDIO MATRIX DECODING 有权
    方法和系统音频矩阵DECODE

    公开(公告)号:EP1362499B1

    公开(公告)日:2012-02-15

    申请号:EP01968271.5

    申请日:2001-08-30

    IPC分类号: H04S3/02

    CPC分类号: H04S3/02

    摘要: A method derives at least three audio signals, each associated with a direction, from two input audio signals. In response to the two input signals, a passive matrix generates a plurality of passive matrix audio signals, including two pairs of passive matrix audio signals, a first pair of passive amtrix audio signals represent directions lying on a first axis and a second pair of passive matrix audio signals represent directions lying on a second axis, the first and second axes being substantially at ninety degrees to ach other. The pairs of passive matrix audio signals are processed to derive a plurality of matrix coefficients therefrom. Th e processing includes deriving a pair of intermediate signals and urging each pair of intermediate signals toward equality in response to a respective error signal. At least three ouput signals are produced by matrix multiplying the two input signals by the matrix coefficients.

    MULTIMODE CODING OF SPEECH-LIKE AND NON-SPEECH-LIKE SIGNALS
    6.
    发明公开
    MULTIMODE CODING OF SPEECH-LIKE AND NON-SPEECH-LIKE SIGNALS 有权
    多模态编码语音喜欢的语言不同SIGNALS

    公开(公告)号:EP2269188A1

    公开(公告)日:2011-01-05

    申请号:EP09720866.4

    申请日:2009-03-12

    IPC分类号: G10L19/08 G10L19/12

    摘要: The invention relates to the coding of audio signals that may include both speech-like and non-speech-like signal components. It describes methods and apparatus for code excited linear prediction (CELP) audio encoding and decoding that employ linear predictive coding (LPC) synthesis filters controlled by LPC parameters, a plurality of codebooks each having codevectors, at least one codebook providing an excitation more appropriate for non-speech-like signals and at least one codebook providing an excitation more appropriate for speech-like signals, and a plurality of gain factors, each associated with a codebook. The encoding methods and apparatus select from the codebooks codevectors and/or associated gain factors by minimizing a measure of the difference between the audio signal and a reconstruction of the audio signal derived from the codebook excitations. The decoding methods and apparatus generate a reconstructed output signal from the LPC parameters, codevectors, and gain factors.

    METHOD FOR DETERMINING UPDATED FILTER COEFFICIENTS OF AN ADAPTIVE FILTER ADAPTED BY AN LMS ALGORITHM WITH PRE-WHITENING
    7.
    发明授权
    METHOD FOR DETERMINING UPDATED FILTER COEFFICIENTS OF AN ADAPTIVE FILTER ADAPTED BY AN LMS ALGORITHM WITH PRE-WHITENING 有权
    用于使用LMS算法的厘定更新后的滤波器系数法调整自适应滤波器与VORWEISSUNG

    公开(公告)号:EP2327156B1

    公开(公告)日:2012-01-18

    申请号:EP09791826.2

    申请日:2009-08-24

    摘要: The application relates to a method for determining at least one updated filter coefficient of an adaptive filter (22) adapted by an LMS algorithm. According to the method, filter coeffi-cients of a first whitening filter (25') are determined, in particular filter coefficients of an LPC whitening filter. The first whitening filter (25') generates a filtered signal. A normaliza-tion value is determined based on one or more computed values obtained in the course of determining the filter coefficients of the first whitening filter (25'). The normalization value is associated with the energy of the filtered signal. At least one updated filter coefficient of the adaptive filter (22) is determined in dependency on the filtered signal and the normaliza-tion value. Preferably, updated filter coefficients for all filter coefficients of the adaptive filter (22) are determined.

    METHOD FOR APPARATUS FOR AUDIO MATRIX DECODING
    8.
    发明公开
    METHOD FOR APPARATUS FOR AUDIO MATRIX DECODING 有权
    方法和系统音频矩阵DECODE

    公开(公告)号:EP1362499A2

    公开(公告)日:2003-11-19

    申请号:EP01968271.5

    申请日:2001-08-30

    IPC分类号: H04S3/02

    CPC分类号: H04S3/02

    摘要: A method derives at least three audio signals, each associated with a direction, from two input audio signals. In response to the two input signals, a passive matrix generates a plurality of passive matrix audio signals, including two pairs of passive matrix audio signals, a first pair of passive amtrix audio signals represent directions lying on a first axis and a second pair of passive matrix audio signals represent directions lying on a second axis, the first and second axes being substantially at ninety degrees to ach other. The pairs of passive matrix audio signals are processed to derive a plurality of matrix coefficients therefrom. Th e processing includes deriving a pair of intermediate signals and urging each pair of intermediate signals toward equality in response to a respective error signal. At least three ouput signals are produced by matrix multiplying the two input signals by the matrix coefficients.