METHOD FOR MODELING SPEECH HARMONIC MAGNITUDES

    公开(公告)号:AU2003216276A1

    公开(公告)日:2003-10-13

    申请号:AU2003216276

    申请日:2003-02-14

    Applicant: MOTOROLA INC

    Abstract: A system or method for modeling a signal, such as a speech signal, in which harmonic frequencies and amplitudes are identified and the harmonic magnitudes are interpolated to obtain spectral magnitudes at a set of fixed frequencies. An inverse transform is applied to the spectral magnitudes to obtain a pseudo auto-correlation sequence, from which linear prediction coefficients are calculated. From the linear prediction coefficients, model harmonic magnitudes are generated by sampling the spectral envelope defined by the linear prediction coefficients. A set of scale factors are then calculated as the ratio of the harmonic magnitudes to the model harmonic magnitudes and interpolated to obtain a second set of scale factors at the set of fixed frequencies. The spectral envelope magnitudes at the set of fixed frequencies are multiplied by the second set of scale factors to obtain new spectral magnitudes and the process is iterated to obtain final linear prediction coefficients. The signal is modeled by the linear prediction coefficients.

    35.
    发明专利
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    公开(公告)号:DK0556354T3

    公开(公告)日:2001-12-17

    申请号:DK92917101

    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.

    MULTI-SEGMENT VECTOR QUANTIZER FOR A SPEECH CODER SUITABLE FOR USE IN A RADIOTELEPHONE

    公开(公告)号:CA2135629C

    公开(公告)日:2000-02-08

    申请号:CA2135629

    申请日:1994-03-07

    Applicant: MOTOROLA INC

    Abstract: A Vector-Sum Excited Linear Predictive Coding (VSELP) speech coder provides improved quality and reduced complexity over a typical speech coder. VSELP uses a codebook which has a predefined structure such that the computations required for the codebook search process can be significantly reduced. This VSELP speech coder uses single or multi-segment vector quantizer of the reflection coefficients based on a Fixed-Point-Lattice-Technique (FLAT). Additionally, this speech coder uses a pre-quantizer to reduce the vector codebook search complexity and a high-resolution scalar quantizer to reduce the amount of memory needed to store the reflection coefficient vector codebooks. Resulting in a high quality speech coder with reduced computations and storage requirements.

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