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    • 1. 发明授权
    • Variable dimension spectral magnitude quantization apparatus and method using predictive and mel-scale binary vector
    • 可变维频谱幅度量化装置和方法,使用预测和梅尔二进制向量
    • US06606592B1
    • 2003-08-12
    • US09584107
    • 2000-05-31
    • Yong-duk ChoMoo-young Kim
    • Yong-duk ChoMoo-young Kim
    • G10L1906
    • G10L19/06G10L25/12G10L2019/0007
    • A variable dimension spectral magnitude quantization apparatus and method using a predictive and mel scale binary vector is provided. The apparatus according to linear prediction spectral envelope and residual spectral envelope quantization using low order linear prediction modeling and residual spectrum modeling, includes a predictive quantizer for obtaining a predictive-quantized first residual spectral envelope from a quantized previous residual spectral envelope, a mel-scale binary vector quantizer for obtaining a second residual spectral envelope represented with a linear scale code vector using a mel-scale binary vector codebook, a synthesized spectral envelope generator for adding the output of the predictive quantizer and the output of the mel-scale binary vector quantizer to generate a quantized residual spectral envelope and multiplying the quantized residual spectral envelope by a corresponding quantized linear prediction spectral envelope to generate a synthesized spectral envelope, a comparator for comparing the synthesized spectral envelope with an original spectral envelope, and a minimum value detector for detecting a minimum value from the values sequentially obtained by the comparator.
    • 提供了一种使用预测和梅尔标度二进制向量的可变维频谱幅度量化装置和方法。 根据线性预测频谱包络和使用低阶线性预测建模和残余频谱建模的残余频谱包络量化的装置包括预测量化器,用于从量化的先前剩余频谱包络获得预测量化的第一残余频谱包络, 二进制矢量量化器,用于获得使用梅尔二进制矢量码本的线性比例代码矢量表示的第二残差频谱包络;合成频谱包络发生器,用于将预测量化器的输出和色标二进制矢量量化器的输出相加 产生量化的残余频谱包络并将量化的残余频谱包络乘以对应的量化的线性预测频谱包络以产生合成频谱包络,用于将合成频谱包络与原始频谱包络比较的比较器和最小值检测 r用于从由比较器顺序获得的值中检测最小值。
    • 2. 发明授权
    • Method and apparatus for determining multiband voicing levels using frequency shifting method in vocoder
    • 在声码器中使用频移方法确定多频带发声级别的方法和装置
    • US06233551B1
    • 2001-05-15
    • US09296242
    • 1999-04-22
    • Yong-duk ChoMoo-young Kim
    • Yong-duk ChoMoo-young Kim
    • G10L1106
    • G10L25/93G10L2025/937
    • A method and an apparatus for determining multiband voicing levels using a frequency moving method in a vocoder are provided. The method for determining the multiband voicing levels using the frequency moving method according to the present invention in the vocoder includes the steps of (a) applying a window to an input voice signal and obtaining a power spectrum from a voice spectrum obtained by Fourier converting a windowed signal, (b) moving the frequency of each subband to an origin after dividing the power spectrum into a predetermined number of subbands, (c) obtaining autocorrelation values of the respective subbands by inverse Fourier converting the power spectrum the frequency of which is moved to the origin, and (d) normalizing the respective autocorrelation values and determining the voicing levels of the subbands from the normalized autocorrelation values.
    • 提供了一种使用声码器中的频率移动方法来确定多频带发声级别的方法和装置。 在声码器中使用根据本发明的频率移动方法来确定多频带发声级别的方法包括以下步骤:(a)将窗口应用于输入语音信号,并从通过傅里叶变换获得的语音频谱获得功率谱 (b)将功率谱分割成预定数量的子带之后,将每个子带的频率移动到原点,(c)通过傅立叶逆变换其频率被移动的功率谱来获得各个子带的自相关值 并且(d)归一化各自自相关值并根据归一化的自相关值确定子带的发声电平。
    • 3. 发明授权
    • Apparatus for quantizing spectral envelope including error selector for selecting a codebook index of a quantized LSF having a smaller error value and method therefor
    • 用于量化频谱包络的​​装置,包括用于选择具有较小误差值的量化LSF的码本索引的误差选择器及其方法
    • US06275796B1
    • 2001-08-14
    • US09060345
    • 1998-04-15
    • Moo-young KimYong-duk ChoHong-kook Kim
    • Moo-young KimYong-duk ChoHong-kook Kim
    • G10L2100
    • G10L19/07
    • An apparatus for quantizing a spectral envelope with noise robustness showing high performance even under a background noise environment and a channel noise environment, and a method therefor, are provided. The spectral envelope quantizing apparatus includes a spectral envelope quantizing apparatus with noise robustness for representing a spectral envelope of speech by a minimum number of bits for the optimal coding of a speech signal. The apparatus includes a line spectrum frequencies (LSFs) input portion for converting linear predictive coding coefficients extracted from the speech into Nth order line spectrum frequencies coefficients and inputting the coefficients as the LSFs of a current frame. It also includes a linked split-vector quantizing portion for dividing the LSFs into a predetermined number of linked sub-vectors and quantizing the sub-vectors, and a predictive linked split-vector quantizing portion for obtaining the difference between the LSFs and the LSFs of a previous frame and vector-quantizing the difference. The apparatus further includes an error selector for comparing the error values of the LSFs quantized in the linked split-vector quantizing portion and the predictive linked split-vector quantizing portion, selecting the codebook index of the quantized LSFs having the smaller error value, and outputting the selected codebook index together with a mode bit.
    • 提供即使在背景噪声环境和信道噪声环境下也显示高性能的具有噪声鲁棒性的频谱包络的​​量化装置及其方法。 频谱包络量化装置包括具有噪声鲁棒性的频谱包络量化装置,用于表示用于语音信号的最佳编码的最小位数的语音的频谱包络。 该装置包括用于将从语音提取的线性预测编码系数转换为第N行频谱频率系数的线谱频率(LSF)输入部分,并输入系数作为当前帧的LSF。 它还包括用于将LSF划分成预定数量的链接子向量并量化子向量的链接分割矢量量化部分,以及用于获得LSF和LSF之间的差异的预测链接分割矢量量化部分 前一帧和向量量化差值。 该装置还包括:误差选择器,用于比较链接的分裂矢量量化部分中量化的LSF的误差值与预测链接的分裂矢量量化部分,选择具有较小误差值的量化LSF的码本索引,并输出 所选码本索引以及模式位。
    • 4. 发明授权
    • Pitch determination apparatus and method using spectro-temporal autocorrelation
    • 使用频谱自相关的音调确定装置和方法
    • US06208958B1
    • 2001-03-27
    • US09226115
    • 1999-01-07
    • Yong-duk ChoMoo-Young Kim
    • Yong-duk ChoMoo-Young Kim
    • G10L1302
    • G10L25/90G10L25/06
    • A pitch determination apparatus and method using spectro-temporal autocorrelation to prevent pitch determination errors are provided. The pitch determination apparatus using spectro-temporal autocorrelation includes a formant bandwidth extension unit for extending a formant bandwidth to reduce the influence of the first formant with respect to an input voice, a temporal autocorrelation calculation unit for calculating an autocorrelation value of a time axial voice within a candidate pitch range with respect to a time axial speech signal output from the formant bandwidth extension unit, a spectral autocorrelation calculation unit for transforming the time axial speech signal output from the formant bandwidth extension unit into a frequency axial signal, and calculating an autocorrelation value between frequency axis amplitude spectrums within the candidate pitch range, an autocorrelation value synthesis unit for summing the autocorrelation values obtained by the temporal and spectral autocorrelation calculation units and obtaining a spectro-temporal autocorrelation value, and a pitch determination unit for determining a pitch having a maximum spectro-temporal autocorrelation value as a final pitch. According to this apparatus, pitch determination errors are reduced by determining a pitch using the temporal and spectral autocorrelation values, thus improving the quality of speech communication.
    • 提供了使用频谱自相关以防止音调确定误差的音调确定装置和方法。 使用频谱自相关的音调确定装置包括:共振峰带宽扩展单元,用于扩展共振峰带宽以减少第一共振峰相对于输入声音的影响;时间自相关计算单元,用于计算时间轴向声音的自相关值 在从共振峰带宽扩展单元输出的时间轴向语音信号的候选音调范围内,频谱自相关计算单元,用于将从共振峰带宽扩展单元输出的时间轴语音信号变换为频率轴向信号,并计算自相关 在候选音调范围内的频率轴振幅谱之间的值;自相关值合成单元,用于对由时间和频谱自相关计算单元获得的自相关值求和并获得谱时间自相关值;以及音调确定单元,用于确定 具有作为最终间距的最大频谱自相关值的音调。 根据该装置,通过使用时间和频谱自相关值确定音调来减小音调确定误差,从而提高语音通信的质量。
    • 5. 发明授权
    • Pitch estimation method for a low delay multiband excitation vocoder
allowing the removal of pitch error without using a pitch tracking
method
    • 用于低延迟多频激励声码器的音调估计方法,允许在不使用音调跟踪方法的情况下去除音调误差
    • US6119081A
    • 2000-09-12
    • US148777
    • 1998-09-04
    • Yong-duk ChoMoo-young Kim
    • Yong-duk ChoMoo-young Kim
    • G10L19/125G10L25/90G10L21/00
    • G10L19/125G10L25/90
    • A of estimating a pitch in a multiband excitation vocoder is provided. The method includes the steps of (a) obtaining an error amount with respect to respective pitch candidates in a predetermined pitch area from an input voice magnitude spectrum, (b) obtaining a weighted function with respect to the respective pitch candidates, (c) obtaining a weighted error amount with respect to the respective pitch candidates, and (d) determining the candidate pitch having the minimum error amount in the weighted error amount with respect to the respective pitch candidates to be an estimated pitch. According to the present invention, in the vocoder of the multiband excitation method, it is possible to obtain high speech quality due to a short delay time since it is possible to remove a gross pitch error without using a pitch tracking method.
    • 提供了估计多频激励语音编码器中的音调的A。 该方法包括以下步骤:(a)从输入的语音幅度频谱获得关于预定音调区域中的相应音调候选的误差量,(b)获得关于各音调候选的加权函数,(c)获得 相对于各个音调候选的加权误差量,以及(d)将相对于各音调候选的加权误差量中具有最小误差量的候选音调确定为估计音调。 根据本发明,在多波段激励方法的声码器中,由于可以在不使用音高跟踪方法的情况下去除总音调误差,所以可以获得由于延迟时间短而导致的高语音质量。
    • 6. 发明授权
    • Voice coding and decoding method and device therefor
    • 语音编码和解码方法及其设备
    • US5884251A
    • 1999-03-16
    • US863956
    • 1997-05-27
    • Hong-kook KimYong-duk ChoMoo-young KimSang-ryong Kim
    • Hong-kook KimYong-duk ChoMoo-young KimSang-ryong Kim
    • G10L19/04G10L19/00G10L19/12H03M7/30G10L3/02G10L9/00
    • G10L19/12
    • In a voice coding and decoding method and apparatus using an RCELP technique, a CELP-series decoder can be obtained at a low transmission rate. A voice spectrum is extracted by performing a short-term linear prediction on voice signal. An error range in a formant region is widened during adaptive and renewal codebook search by passing said preprocessed voice through a formant weighting filter and widening an error range in a pitch on-set region by passing the same through a voice synthesis filter and a harmonic noise shaping filter. An adaptive codebook is searched using an open-loop pitch extracted on the basis of the residual minus of a speech. A renewal excited codebook produced from an adaptive codebook excited signal is searched. Finally, a predetermined bit is allocated to various parameters to form a bit stream.
    • 在使用RCELP技术的语音编码和解码方法和装置中,可以以低传输速率获得CELP系列解码器。 通过对语音信号执行短期线性预测来提取语音频谱。 在自适应和更新码本搜索期间,通过使所述预处理语音通过共振峰加权滤波器并在音调设定区域中扩大误差范围,通过使其通过语音合成滤波器和谐波噪声来加宽共振峰区域中的误差范围 成型过滤器。 使用基于语音的残差减去提取的开环音调来搜索自适应码本。 搜索由自适应码本激励信号产生的更新兴奋码本。 最后,将预定比特分配给各种参数以形成比特流。