DIGITAL SPEECH CODER HAVING IMPROVED SUB-SAMPLE RESOLUTION LONG-TERM PREDICTOR
    11.
    发明公开
    DIGITAL SPEECH CODER HAVING IMPROVED SUB-SAMPLE RESOLUTION LONG-TERM PREDICTOR 失效
    具有改进的子样品分辨率长期预测的数字语音编码器

    公开(公告)号:EP0450064A4

    公开(公告)日:1995-04-05

    申请号:EP91905041

    申请日:1990-06-25

    Applicant: MOTOROLA INC

    CPC classification number: G10L19/12

    Abstract: A digital speech coder includes a long-term filter (124) having an improved sub-sample resolution long-term predictor which allows for subsample resolution for the lag parameter L. A frame of N samples of input speech vector s(n) is applied to an adder (510). The output of the adder (510) produces the output vector b(n) for the long term filter (124). The output vector b(n) is fed back to a delayed vector generator block (530) of the long-term predictor. The nominal long-term predictor lag parameter L is also input to the delayed vector generator block (530). The long-term predictor lag parameter L can take on non-integer values, which may be multiples of one half, one third, one fourth or any other rational fraction. The delayed vector generator (530) includes a memory which holds past samples of b(n). In addition, interpolated samples of b(n) are also calculated by the delayed vector generator (530) and stored in its memory, at least one interpolated sample being calculated and stored between each past sample of b(n). The delayed vector generator (530) provides output vector q(n) to the long-term multiplier block (520), which scales the long-term predictor response by the long-term predictor coefficient beta . The scaled output beta q(n) is then applied to the adder (510) to complete the feedback loop of the recursive filter (124).

    METHOD AND APPARATUS FOR CODING GAIN INFORMATION IN A SPEECH CODING SYSTEM
    12.
    发明公开
    METHOD AND APPARATUS FOR CODING GAIN INFORMATION IN A SPEECH CODING SYSTEM 审中-公开
    方法和装置进行编码VERSTûRKUNGSINFORMATIONEN在演讲编码系统

    公开(公告)号:EP1563489A4

    公开(公告)日:2007-06-13

    申请号:EP03768792

    申请日:2003-11-06

    Applicant: MOTOROLA INC

    CPC classification number: G10L19/083

    Abstract: A speech coder (300) that performs analysis-by-synthesis coding of a signal determines gain parameters for each constituent component of multiple constituent components of a synthetic excitation signal (ex(n)). The speech coder generates a target vector (p(n)) based on an input signal (s(n)). The speech coder further generates multiple constituent components associated with the synthetic excitation signal, wherein one constituent component of the multiple constituent components is based on a shifted version of another constituent component of the multiple constituent components. The speech coder further evaluates an error criteria based on the target vector and the multiple constituent components to determine a gain associated with each constituent component of the multiple constituent components.

    Abstract translation: 甲语音编码器那样进行分析边合成bestimmt用于合成激励信号的多个构成部件的各构成部件的参数的增益的信号的编码。 语音编码器基因利率在输入信号基于目标矢量。 语音编码器与合成激励信号相关联的进一步的基因速率的多个构成部件,worin所述多个构成部件中的一个构成成分是基于所述多个构成部件的另一组成部件的一个移位版本。 编码器进一步评估语音错误基于目标向量和标准的多个构成部件确定性矿与多个构成组件的每个组成部件相关联的增益。

    METHOD AND APPARATUS FOR SPEECH RECONSTRUCTION IN A DISTRIBUTED SPEECH RECOGNITION SYSTEM
    13.
    发明公开
    METHOD AND APPARATUS FOR SPEECH RECONSTRUCTION IN A DISTRIBUTED SPEECH RECOGNITION SYSTEM 有权
    方法和设备语音重建分布式语音识别系统

    公开(公告)号:EP1395978A4

    公开(公告)日:2005-09-21

    申请号:EP02709089

    申请日:2002-01-18

    Applicant: MOTOROLA INC

    CPC classification number: G10L15/30 G10L19/00 G10L19/093 G10L25/18

    Abstract: A method of reconstructing speech input at a communication device comprises receiving, at the communication device, encoded data that includes encoded spectral data and encoded energy data of the speech input, the encoded spectral data being encoded as a series of mel-frequency cepstral coefficients. The method further comprises decoding, at the communication device, the encoded spectral data and encoded energy data to determine the spectral data and energy data, wherein decoding comprises: performing an inverse discrete cosine transform on the mel-frequency cepstral coefficients at harmonic mel-frequencies corresponding to a pitch period of the speech input to determine log-spectral magnitudes of the speech input at the harmonic mel-frequencies, and exponentiating the log-spectral magnitudes to determine the spectral magnitudes of the speech input. The method also comprises combining the spectral data and energy data to reconstruct the speech input at the communication device. A communication device for use in distributed speech recognition system is also disclosed.

    METODO Y APARATO PARA CALCULAR ENERGIA DE BANDA ALTA EN UN SISTEMA DE EXTENSION DE ANCHO DE BANDA.

    公开(公告)号:MX2010008279A

    公开(公告)日:2010-08-30

    申请号:MX2010008279

    申请日:2009-01-28

    Applicant: MOTOROLA INC

    Abstract: Un método (100) que incluye recibir (101) una señal de audio digital de entrada que comprende una señal de banda estrecha. La señal de audio digital de entrada se procesa (102) para generar una señal procesada de audio digital. Un nivel de energía de banda alta correspondiente a la señal de audio digital de entrada se calcula (103) con base en una energía calculada de una banda de transición de la señal procesada de audio digital en un rango predeterminado de frecuencias superiores de un ancho de banda de banda estrecha. Una señal de audio digital de banda alta se genera (104) con base en el nivel de energía de banda alta y un espectro calculado de banda alta correspondiente al nivel de energía de banda alta.

    19.
    发明专利
    未知

    公开(公告)号:DE602005008974D1

    公开(公告)日:2008-09-25

    申请号:DE602005008974

    申请日:2005-06-06

    Applicant: MOTOROLA INC

    Abstract: A method for equalizing a speech signal generated within a pressurized air delivery system, the method including the steps of: generating an inhalation noise model ( 1152 ) based on inhalation noise; receiving an input signal ( 802 ) that includes a speech signal; and equalizing the speech signal ( 1156 ) based on the noise model.

    20.
    发明专利
    未知

    公开(公告)号:ES2167318T3

    公开(公告)日:2002-05-16

    申请号:ES92917101

    申请日:1992-08-03

    Applicant: MOTOROLA INC

    Abstract: A method for protecting information bits wherein input data bits, at least some of which are to be protected, are sorted based upon information determined from a subset of the input data bits. An error control coding technique is applied to at least some of the sorted bits. In the preferred embodiment, an input data stream of voice coder bits is separated into arrays of bits. A first array (302) comprises voice coder bits needing error protection, with the bits arranged in order of importance determined by voicing mode. The second array (303) comprises bits that will not be error protected. The bits from the first array are provided to the input of an encoder (304), then the encoded bits are combined (305) with the bits from the second array (303) to form a bit stream.

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