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    Method and apparatus for speech encoding and decoding by sinusoidal analysis and waveform encoding with phase reproducibility
    3.
    发明授权
    Method and apparatus for speech encoding and decoding by sinusoidal analysis and waveform encoding with phase reproducibility 失效
    通过正弦分析和具有相位再现性的波形编码进行语音编码和解码的方法和装置

    公开(公告)号：US07454330B1

    公开(公告)日：2008-11-18

    申请号：US08736546

    申请日：1996-10-24

    申请人： Masayuki Nishiguchi ,  Kazuyuki Iijima ,  Jun Matsumoto ,  Shiro Omori

    发明人： Masayuki Nishiguchi ,  Kazuyuki Iijima ,  Jun Matsumoto ,  Shiro Omori

    IPC分类号： G10L19/14

    CPC分类号： G10L19/0212 ,  G10L19/02 ,  G10L19/04 ,  G10L19/06 ,  G10L19/12 ,  G10L25/27 ,  G10L25/93

    摘要： A speech encoding method and apparatus in which an input speech signal is divided in terms of blocks or frames as encoding units and encoded in terms of the encoding units, whereby explosive and fricative consonants can be impeccably reproduced, while there is an attenuation of the occurrence of foreign sounds being generated at a transient portion between voiced (V) and unvoiced (UV) portions, so that the speech with high clarity devoid of “stuffed” feeling may be produced. The encoding apparatus includes a first encoding unit for finding residuals of linear predictive coding (LPC) of an input speech signal for performing harmonic coding and a second encoding unit for encoding the input speech signal by waveform coding. The first encoding unit and the second encoding unit are used for encoding a voiced (V) portion and an unvoiced (UV) portion of the input signal, respectively. Code excited linear prediction (CELP) encoding employing vector quantization by a closed loop search of an optimum vector using an analysis-by-synthesis method is used for the second encoding unit. A corresponding decoding method and apparatus is also provided.

    摘要翻译： 一种语音编码方法和装置，其中输入语音信号以块或帧为单位编码，并以编码单位编码，由此可以无可挑剔地复制爆炸和摩擦辅音，同时存在衰减的发生 在V（V）和无声（UV）部分之间的瞬态部分产生外来声音，从而可能产生具有高“透明度”感的语音。 编码装置包括：第一编码单元，用于求出用于执行谐波编码的输入语音信号的线性预测编码（LPC）的残差;以及第二编码单元，用于通过波形编码对输入的语音信号进行编码。 第一编码单元和第二编码单元分别用于对输入信号的有声（V）部分和无声（UV）部分进行编码。 第二编码单元使用通过使用合成分析法的最佳向量的闭环搜索采用矢量量化的码激励线性预测（CELP）编码。 还提供了相应的解码方法和装置。

    Echo canceling apparatus and method, and voice reproducing apparatus
    4.
    发明授权
    Echo canceling apparatus and method, and voice reproducing apparatus 失效
    回波消除装置和方法以及语音再现装置

    公开(公告)号：US06694018B1

    公开(公告)日：2004-02-17

    申请号：US09422249

    申请日：1999-10-21

    申请人： Shiro Omori

    发明人： Shiro Omori

    IPC分类号： H04M908

    CPC分类号： H04M9/082

    摘要： An echo canceller is provided in which a down-sampling circuit converts a 16-kHz sampling frequency of a wide-band voice signal output to an 8-kHz sampling frequency of a narrow-band voice signal supplied at an input terminal, an adaptive filter estimates, from the wide-band voice signal whose sampling frequency has been down-sampled to 8 kHz in the down-sampling circuit, an echo signal coming from a speaker to a microphone and having an echo path characteristic imparted to the echo signal by an echo path filter, and a subtraction circuit subtracts from the microphone input signal the echo signal having been estimated by the adaptive filter.

    摘要翻译： 提供了一种回波消除器，其中下采样电路将宽带语音信号输出的16kHz采样频率转换为在输入端提供的窄带语音信号的8kHz采样频率，自适应滤波器 从下采样电路中采样频率已经被采样到8kHz的宽带语音信号，从扬声器到麦克风的回波信号估计出来，具有通过一个回波信号赋予回波信号的回波路径特性 回波路径滤波器和减法电路从麦克风输入信号中减去由自适应滤波器估计的回波信号。

    Signal band expanding method and apparatus and signal synthesis method and apparatus
    5.
    发明授权
    Signal band expanding method and apparatus and signal synthesis method and apparatus 失效
    信号带扩展方法及装置及信号合成方法及装置

    公开(公告)号：US06539355B1

    公开(公告)日：2003-03-25

    申请号：US09417585

    申请日：1999-10-14

    申请人： Shiro Omori ,  Masayuki Nishiguchi

    发明人： Shiro Omori ,  Masayuki Nishiguchi

    IPC分类号： G10L1902

    CPC分类号： G10L21/038

    摘要： A bandwidth expanding method and apparatus in which frequency characteristics of high-frequency components of broad band signals can be adjusted to the liking of the user, overflow due to addition is prevented from occurring without power variations being perceived by a user, the number of broad band formants is reduced, and emphasis is attached to the rough structure of the spectrum, so that the produced broad band speech signals can be improved in quality. To this end, in a speech bandwidth expansion device, frequency characteristics of the frequency components not less than 3400 Hz are adjusted by preset alterable parameter values and summed to the original narrow band speech components. If overflow has occurred in a sample, the high-range gain of the sample is lowered to a level below the overflow level before proceeding to addition. Also, broad band autocorrelation &ggr;w is generated and inverse-transformed in an inverse parameter conversion unit to produce broad band linear prediction coefficient &agr;W to synthesize the broad-band speech in a linear predictive coding synthesis unit.

    摘要翻译： 宽带信号的高频分量的频率特性可以根据用户的喜好进行调整的带宽扩展方法和装置，防止由于添加而导致的溢出，而不会由用户感知到功率变化，广泛的数量 频带共振峰减少，重点在于光谱的粗糙结构，从而可以提高产生的宽带语音信号的质量。 为此，在语音带宽扩展装置中，频率分量不小于3400Hz的频率特性通过预设的可变参数值进行调整，并与原始窄带语音分量相加。 如果在样品中发生溢出，则在继续添加之前，将样品的高范围增益降低到低于溢出水平的水平。 此外，在逆参数转换单元中产生宽带自相关法拉姆并逆变换，以产生宽带线性预测系数αW，以在线性预测编码合成单元中合成宽带语音。

    Information processing apparatus and method, and recording medium
    7.
    发明授权
    Information processing apparatus and method, and recording medium 有权
    信息处理装置和方法以及记录介质

    公开(公告)号：US06711538B1

    公开(公告)日：2004-03-23

    申请号：US09672907

    申请日：2000-09-28

    申请人： Shiro Omori ,  Masayuki Nishiguchi

    发明人： Shiro Omori ,  Masayuki Nishiguchi

    IPC分类号： G10L1910

    CPC分类号： G10L21/038

    摘要： In order to improve the accuracy of an excitation source for a band-spreading apparatus and to generate a wide-band signal having no gaps, an &agr; band-widening section generates a prediction coefficient &agr;W of a wide-band speech signal from a prediction coefficient &agr;N of a narrow-band speech signal. An oversampling apparatus oversamples a narrow-band speech signal sndN. An interpolation section generates an adaptive signal excPW of a wide-band speech signal from an adaptive signal excPN of the narrow-band speech signal. A zero-filling section generates a noise signal of a wide-band speech signal from a noise signal excNN of the narrow-band speech signal. A noise addition section adds a noise signal which is a gap of the wide-band speech signal and generates a noise signal excNW. An adder generates an excitation source excPW for the wide-band speech signal from the adaptive signal excPW and the noise signal excNW of the wide-band speech signal. A wide-band LPC combining section generates a wide-band speech signal. A band suppression section suppresses a frequency band contained in the narrow-band speech signal within the wide-band speech signal. An adder outputs a wide-band speech signal sndW from the wide-band speech signal and the oversampled narrow-band speech signal.

    摘要翻译： 为了提高频带扩展装置的激励源的精度，并且生成没有间隙的宽带信号，α带宽部分从预测系数产生宽带语音信号的预测系数αW 窄带语音信号的αN。 过采样装置对窄带语音信号sndN进行抽样。 内插部分从窄带语音信号的自适应信号excPN生成宽带语音信号的自适应信号excPW。 零填充部分从窄带语音信号的噪声信号excNN产生宽带语音信号的噪声信号。 噪声添加部分添加作为宽带语音信号的间隙的噪声信号，并产生噪声信号excNW。 加法器从自适应信号excPW和宽带语音信号的噪声信号excNW产生用于宽带语音信号的激励源excPW。 宽带LPC组合部分产生宽带语音信号。 频带抑制部抑制宽带语音信号内的窄带语音信号中包含的频带。 加法器从宽带语音信号和过采样窄带语音信号输出宽带语音信号sndW。

    Voiced/unvoiced decision using a plurality of sigmoid-transformed parameters for speech coding
    8.
    发明授权
    Voiced/unvoiced decision using a plurality of sigmoid-transformed parameters for speech coding 失效
    使用多个S形变换参数进行语音编码的发声/清音决定

    公开(公告)号：US06023671A

    公开(公告)日：2000-02-08

    申请号：US833970

    申请日：1997-04-11

    申请人： Kazuyuki Iijima ,  Masayuki Nishiguchi ,  Jun Matsumoto ,  Shiro Omori

    发明人： Kazuyuki Iijima ,  Masayuki Nishiguchi ,  Jun Matsumoto ,  Shiro Omori

    IPC分类号： G10L11/00 ,  G10L11/02 ,  G10L11/06 ,  G10L15/02 ,  G10L19/00 ,  H03M7/30 ,  G10L9/00

    CPC分类号： G10L25/93

    摘要： A method and apparatus for voiced/unvoiced decision for judging whether an input speech signal is voiced or unvoiced. The input parameters for performing the voiced/unvoiced (V/UV) decision are comprehensively judged in order to enable high-precision V/UV decision by a simplified algorithm. Parameters for the voiced/unvoiced (V/UV) decision include the frame-averaged energy of the input speech signal lev, the normalized autocorrelation peak value r0r, the spectral similarity degree pos, the number of zero crossings nZero, and the pitch lag pch. If these parameters are denoted by x, these parameters are converted by function calculation circuits using a sigmoid function g(x) represented byg(x)=A/(1+exp (-(x-b)/a))where A, a, and b are constants differing with each input parameter. Using the parameters converted by this sigmoid function g(x), the voiced/unvoiced decision is made a V/UV decision circuit.

    摘要翻译： 用于用于判断输入语音信号是有声还是无声的有声/无声决定的方法和装置。 综合判断用于执行有声/无声（V / UV）判定的输入参数，以便通过简化算法实现高精度V / UV判定。 有声/无声（V / UV）决定的参数包括输入语音信号lev的帧平均能量，归一化自相关峰值r0r，频谱相似度pos，过零次数nZero和音调滞后pch 。 如果这些参数由x表示，这些参数由函数计算电路使用由g（x）= A /（1 + exp（ - （xb）/ a））表示的S形函数g（x）转换，其中A，a， b是与每个输入参数不同的常数。 使用由该S形函数g（x）转换的参数，将有声/无声决定作为V / UV判定电路。

    Method and apparatus for decoding and changing the pitch of an encoded speech signal
    9.
    发明授权
    Method and apparatus for decoding and changing the pitch of an encoded speech signal 失效
    用于对编码语音信号进行解码和改变音调的方法和装置

    公开(公告)号：US5873059A

    公开(公告)日：1999-02-16

    申请号：US736989

    申请日：1996-10-25

    申请人： Kazuyuki Iijima ,  Masayuki Nishiguchi ,  Jun Matsumoto ,  Shiro Omori

    发明人： Kazuyuki Iijima ,  Masayuki Nishiguchi ,  Jun Matsumoto ,  Shiro Omori

    IPC分类号： G10L11/06 ,  G10L13/00 ,  G10L13/02 ,  G10L19/04 ,  G10L21/04 ,  H03M7/30 ,  G10L9/00

    CPC分类号： G10L13/033 ,  G10L21/01 ,  G10L19/087

    摘要： A method and apparatus for reproducing speech signals at a controlled speed and for synthesizing speech includes a dividing unit that divides the input speech into time segments and an encoding unit that discriminates whether each of the speech segments is voiced or unvoiced. Based on the results of the discrimination, the encoding unit performs sinusoidal synthesis and encoding for voiced segments and vector quantization by closed-loop search for an optimum vector using an analysis-by-synthesis method for unvoiced segments in order to find encoded parameters. A period modification unit modifies the length of time associated with each signal segment and calculates a set of modified encoded parameters. In the speech synthesizing unit, encoded speech signal data is output from the encoding unit and pitch data and amplitude data specifying the spectral envelope are sent via a data conversion unit to a waveform synthesis unit, where the number of amplitude data points of the spectral envelope is changed without changing the shape of the spectral envelope, so that the pitch of the signal may be varied without changing its phoneme. A waveform synthesis unit synthesizes the speech waveform based on the converted spectral envelope data and pitch data.

    摘要翻译： 用于以受控速度再现语音信号并用于合成语音的方法和装置包括将输入语音划分成时间段的分割单元和鉴别每个语音段是有声还是无声的编码单元。 基于鉴别的结果，编码单元通过使用用于清音段的合成分析方法对最佳向量进行闭环搜索，对浊音段和矢量量化进行正弦合成和编码，以便找到编码参数。 周期修改单元修改与每个信号段相关联的时间长度，并计算一组经修改的编码参数。 在语音合成单元中，编码语音信号数据从编码单元输出，音调数据和指定频谱包络的​​振幅数据经由数据转换单元发送到波形合成单元，其中频谱包络的​​振幅数据点的数量 在不改变频谱包络的​​形状的情况下改变，使得信号的音调可以改变而不改变其音素。 波形合成单元基于转换的频谱包络数据和音调数据来合成语音波形。

    Apparatus and method for encoding/decoding a speech signal using adaptively changing codebook vectors
    10.
    发明授权
    Apparatus and method for encoding/decoding a speech signal using adaptively changing codebook vectors 失效
    使用自适应变化的码本矢量对语音信号进行编码/解码的装置和方法

    公开(公告)号：US5828996A

    公开(公告)日：1998-10-27

    申请号：US736988

    申请日：1996-10-25

    申请人： Kazuyuki Iijima ,  Masayuki Nishiguchi ,  Jun Matsumoto ,  Shiro Omori

    发明人： Kazuyuki Iijima ,  Masayuki Nishiguchi ,  Jun Matsumoto ,  Shiro Omori

    IPC分类号： G10L19/08 ,  G10L19/04 ,  G10L19/12 ,  H03M7/30 ,  G10L7/02

    CPC分类号： G10L19/04 ,  G10L19/12

    摘要： An encoding apparatus in which an input speech signal is divided into blocks and encoded in units of blocks. The encoding apparatus includes an encoding unit for performing CELP encoding having a noise codebook memory containing having codebook vectors generated by clipping Gaussian noise and codebook vectors obtained by learning using the code vectors generated by clipping the Gaussian noise as initial values. The encoding apparatus enables optimum encoding for a variety of speech configurations.

    摘要翻译： 一种编码装置，其中输入语音信号被分成块并以块为单位编码。 该编码装置包括编码单元，用于执行CELP编码，该编码单元具有噪声码本存储器，该噪声码本存储器包含通过使用通过限幅高斯噪声产生的代码矢量进行学习而获得的通过削波高斯噪声和码本矢量生成的码本矢量作为初始值。 编码装置能够对各种语音配置进行最佳编码。