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    • 2. 发明授权
    • Speech coding by code-edited linear prediction
    • 通过编码线性预测的语音编码
    • US5787391A
    • 1998-07-28
    • US658303
    • 1996-06-05
    • Takehiro MoriyaAkitoshi KataokaKazunori ManoSatoshi MikiHitoshi OmuroShinji Hayashi
    • Takehiro MoriyaAkitoshi KataokaKazunori ManoSatoshi MikiHitoshi OmuroShinji Hayashi
    • G10L19/005G10L19/06G10L19/07G10L19/08G10L19/083G10L19/12G10L19/135G10L3/02
    • G10L19/08G10L19/005G10L19/06G10L19/07G10L19/083G10L19/12G10L19/135
    • In a speech coding method of the present invention, initially, a plurality of samples of speech data are analyzed by a linear prediction analysis and thereby prediction coefficients are calculated. Then, the prediction coefficients are quantized, and the quantized prediction coefficients are set in a synthesis filter. Moreover, a pitch period vector is selected from an adaptive codebook in which a plurality of pitch period vectors are stored, and the selected pitch period vector is multiplied by a first gain which is obtained, at the same time, with a second gain. In addition, a noise waveform vector is selected from a random codebook in which a plurality of the noise waveform vectors are stored, and is multiplied by a predicted gain and the second gain. Then, the speech vector is synthesized by exciting the synthesis filter with the pitch period vector multiplied by the first gain, and with the noise waveform vector multiplied by the predicted gain and the second gain. Consequently, speech data comprising a plurality of samples are coded as a unit of a frame operation. Furthermore, the predicted gain multiplied by the noise waveform vector which is selected in a subsequent frame operation, is predicted based on the current noise waveform vector which is multiplied by the predicted gain and the second gain at the current frame operation, and also the previous waveform vector which is multiplied by the predicted gain and the second gain in the previous frame operation.
    • 在本发明的语音编码方法中,首先,通过线性预测分析来分析多个语音数据样本,从而计算出预测系数。 然后,量化预测系数,并将量化的预测系数设置在合成滤波器中。 此外,从存储多个音调周期矢量的自适应码本中选择音调周期矢量,并且将所选择的音调周期矢量乘以与第二增益同时获得的第一增益。 此外,从存储多个噪声波形向量的随机码本中选择噪声波形向量,并将其乘以预测的增益和第二增益。 然后,通过利用乘以第一增益的音调周期矢量并且噪声波形向量乘以预测增益和第二增益来激励合成滤波器来合成语音向量。 因此,包括多个样本的语音数据被编码为帧操作的单位。 此外,基于在当前帧操作中乘以预测增益和第二增益的当前噪声波形向量来预测在后续帧操作中选择的噪声波形向量的预测增益,以及前一帧 在前一帧操作中乘以预测增益和第二增益的波形向量。
    • 4. 发明授权
    • Acoustic signal transform coding method and decoding method having a
high efficiency envelope flattening method therein
    • 声信号变换编码方法及其中具有高效率包络平滑化方法的解码方法
    • US5684920A
    • 1997-11-04
    • US402660
    • 1995-03-13
    • Naoki IwakamiTakehiro MoriyaSatoshi Miki
    • Naoki IwakamiTakehiro MoriyaSatoshi Miki
    • G01R23/16G10L19/02G10L25/12G10L25/27G10L9/16
    • G10L19/0212G10L19/0204G10L25/12G10L25/27
    • An input acoustic signal is subjected to modified discrete cosine transform processing to obtain its spectrum characteristics. Linear prediction coefficients are derived from the input acoustic signal in a linear prediction coding analysis part, and the prediction coefficients are subjected to Fourier transform in a spectrum envelope calculation part to obtain the envelope of the spectrum characteristics of the input acoustic signal. In a normalization part the spectrum characteristics are normalized by the envelope thereof to obtain residual coefficients. Another normalization part normalizes the residual coefficients by a residual-coefficients envelope predicted in a residual-coefficients envelope calculation part, thereby obtaining fine structure coefficients, which are vector-quantized in a quantization part. A de-normalization part de-normalizes the quantized fine structure coefficients. The residual-coefficients envelope calculation part uses the reproduced residual coefficients to predict the envelope of residual coefficients of the subsequent frame.
    • 对输入声信号进行修正的离散余弦变换处理以获得其频谱特性。 线性预测系数从线性预测编码分析部中的输入声信号导出,并且在频谱包络计算部中对预测系数进行傅里叶变换,以获得输入声信号的频谱特性的包络。 在归一化部分中,光谱特性通过其包络进行归一化以获得残差系数。 另一个归一化部分通过在残差系数包络计算部分中预测的残差系数包络对残差系数进行归一化,从而获得在量化部分中矢量量化的精细结构系数。 去归一化部分使量化的精细结构系数解规范化。 剩余系数包络计算部分使用再现的残差系数来预测后续帧的残差系数的包络。