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1. 发明授权

US5819213A Speech encoding and decoding with pitch filter range unrestricted by codebook range and preselecting, then increasing, search candidates from linear overlap codebooks 失效
标题翻译：语音编码和解码与音调滤波器范围不受码本范围和预选择限制，然后增加从线性重叠码本搜索候选
公开(公告)号：US5819213A
公开(公告)日：1998-10-06
申请号：US791741
申请日：1997-01-30
申请人： Masahiro Oshikiri , Tadashi Amada , Masami Akamine , Kimio Miseki
发明人： Masahiro Oshikiri , Tadashi Amada , Masami Akamine , Kimio Miseki
IPC分类号： G10L19/08 , G10L9/14
CPC分类号： G10L19/08
摘要： A speech encoding method and apparatus including analyzing, using a codebook expressing speech parameters within a predetermined search range, an input speech signal in an audibility weighting filter corresponding to a pitch period longer than the search range of the codebook, and searching, from the codebook, on the basis of the analysis result, a combination of speech parameters by which the distortion of the input speech signal is minimized, and encoding the combination. The apparatus uses an adaptive codebook of pitch and a noise codebook. The codebooks search a group formed by extracting vectors of predetermined length from one original code vector, while sequentially shifting position so that the vectors overlap each other. The search group is further restricted and another preselection is made before the final search. Search is based on inversely convoluted, orthogonally transformed vectors.
摘要翻译：一种语音编码方法和装置，包括使用在预定搜索范围内表达语音参数的码本来分析与音码周期相对应的音频周期的声音加权滤波器中的输入语音信号，并且从码本中搜索基于分析结果，将输入语音信号的失真最小化的语音参数的组合以及编码该组合。该装置使用音调和噪声码本的自适应码本。码本搜索通过从一个原始码矢量提取预定长度的矢量而形成的组，同时依次移位位置使得矢量彼此重叠。搜索组进一步限制，并在最终搜索之前进行另一预选。搜索基于逆卷积正交变换载体。

2. 发明授权

US06704702B2 Speech encoding method, apparatus and program 失效
标题翻译：语音编码方法，装置和程序
公开(公告)号：US06704702B2
公开(公告)日：2004-03-09
申请号：US09726562
申请日：2000-12-01
申请人： Masahiro Oshikiri , Kimio Miseki , Masami Akamine
发明人： Masahiro Oshikiri , Kimio Miseki , Masami Akamine
IPC分类号： G10L1104
CPC分类号： G10L25/93 , G10L19/09 , G10L25/78
摘要： A speech encoding method, apparatus and program wherein an input speech signal is divided into a plurality of frames each having a predetermined length, each of the frames is subdivided into a plurality of subframes, a predictive pitch period of a subframe in a to-be-encoded current frame is obtained by using pitch periods of at least two frames of the current frame and past and future frames with respect to the current frame; a pitch period of a subframe in the current frame is obtained by using the predictive pitch period, a relative pitch pattern codebook storing a plurality of relative pitch patterns representing fluctuations in pitch periods of a plurality of subframes is prepared, and a change in pitch period of plural subframes is expressed with one relative pitch pattern selected from the relative pitch pattern codebook.
摘要翻译：一种语音编码方法，装置和程序，其中输入语音信号被划分为具有预定长度的多个帧，每个帧被细分为多个子帧，子帧的预测音调周期通过使用当前帧的至少两帧的音调周期和相对于当前帧的过去和未来帧来获得编码的当前帧; 通过使用预测音调周期来获得当前帧中的子帧的音调周期，准备存储表示多个子帧的音调周期的波动的多个相对音调模式的相对音调模式码本，并且音调周期的变化由相对的音调图案码本中选择的一个相对音调图形来表示多个子帧。

3. 发明授权

US6167375A Method for encoding and decoding a speech signal including background noise 失效
标题翻译：用于对包括背景噪声的语音信号进行编码和解码的方法
公开(公告)号：US6167375A
公开(公告)日：2000-12-26
申请号：US39317
申请日：1998-03-16
申请人： Kimio Miseki , Masahiro Oshikiri , Tadashi Amada , Masami Akamine
发明人： Kimio Miseki , Masahiro Oshikiri , Tadashi Amada , Masami Akamine
IPC分类号： G10L19/00 , G10L19/02
CPC分类号： G10L19/012 , G10L19/02
摘要： A method for encoding speech wherein an input speech signal is separated by a component separator into a first component mainly constituted by speech and a second component mainly constituted by a background noise at each predetermined unit of time, a bit allocation selector selects bit allocation for each component based on the first and second components from among a plurality of predetermined candidates for bit allocation, a speech encoder and a noise encoder encode the first and second components from the component separator based on the bit allocation according to predetermined different methods for encoding, and a multiplexer multiplexes encoded data of the first and second components and information on the bit allocation and outputs them as transmitted encoded data.
摘要翻译：一种编码语音的方法，其中输入语音信号由分量分离器分离成主要由语音构成的第一分量和主要由每个预定单位时间的背景噪声构成的第二分量，比特分配选择器选择每个基于来自用于比特分配的多个预定候选中的第一和第二分量的分量，语音编码器和噪声编码器根据预定的不同编码方法基于比特分配对来自分量分离器的第一和第二分量进行编码，以及复用器复用第一和第二分量的编码数据和关于比特分配的信息，并将其作为发送的编码数据输出。

4. 发明授权

US06202046B1 Background noise/speech classification method 失效
标题翻译：背景噪音/语音分类方法
公开(公告)号：US06202046B1
公开(公告)日：2001-03-13
申请号：US09012792
申请日：1998-01-23
申请人： Masahiro Oshikiri , Kimio Miseki , Masami Akamine
发明人： Masahiro Oshikiri , Kimio Miseki , Masami Akamine
IPC分类号： G10L1106
CPC分类号： G10L25/93 , G10L19/09 , G10L25/78
摘要： In a background noise/speech classification method, whether a digital input signal input through an input terminal is background noise or speech is decided by a background noise/speech decision section on the basis of calculated frame power and a calculated LSP coefficient which are obtained by supplying the input signal to a feature amount calculation section and estimated frame power and an estimated LSP coefficient obtained by an estimated feature amount update section. Thereafter, the estimated feature amount update section updates the estimated frame power and the estimated LSP coefficient by using the frame power and the LSP coefficient obtained by the feature amount calculation section to prepare for the next frame.
摘要翻译：在背景噪声/语音分类方法中，基于计算出的帧功率和计算出的LSP系数，通过背景噪声/语音判定部分确定通过输入端子输入的数字输入信号是背景噪声还是语音是由将输入信号提供给特征量计算部分，并且估计帧功率和由估计特征量更新部分获得的估计LSP系数。此后，估计特征量更新部分通过使用由特征量计算部分获得的帧功率和LSP系数来准备下一帧来更新估计的帧功率和估计的LSP系数。

5. 发明授权

US5864798A Method and apparatus for adjusting a spectrum shape of a speech signal 失效
标题翻译：用于调整语音信号的频谱形状的方法和装置
公开(公告)号：US5864798A
公开(公告)日：1999-01-26
申请号：US714260
申请日：1996-09-17
申请人： Kimio Miseki , Masahiro Oshikiri , Akinobu Yamashita , Masami Akamine , Tadashi Amada
发明人： Kimio Miseki , Masahiro Oshikiri , Akinobu Yamashita , Masami Akamine , Tadashi Amada
IPC分类号： G10L19/14 , G10L9/00
CPC分类号： G10L19/26 , G10L25/12
摘要： Adjusting the shape of a spectrum of a speech signal includes the steps of using a first filter with pole-zero transfer function A(z)/B(z) for subjecting a speech signal to a spectrum envelop emphasis and a second filter cascade-connected with the first filter, for compensating for a spectral tilt due to the first filter, independently deriving two filter coefficients used in the second filter for compensating for the spectral tilt from the pole-zero transfer function, and compensating for the spectral tilt corresponding to the pole-zero transfer function according to the derived filter coefficients.
摘要翻译：调整语音信号的频谱的形状包括以下步骤：使用具有极零传递函数A（z）/ B（z）的第一滤波器来对语音信号进行频谱包络加强，并且第二滤波器级联连接利用第一滤波器，用于补偿由于第一滤波器引起的频谱倾斜，独立地导出在第二滤波器中使用的两个滤波器系数，用于补偿来自极 - 零传递函数的频谱倾斜，以及补偿对应于根据派生滤波系数的极零传递函数。

6. 发明授权

US5677986A Vector quantizing apparatus 失效
标题翻译：矢量量化装置
公开(公告)号：US5677986A
公开(公告)日：1997-10-14
申请号：US451174
申请日：1995-05-26
申请人： Tadashi Amada , Kimio Miseki , Masami Akamine , Masahiro Oshikiri
发明人： Tadashi Amada , Kimio Miseki , Masami Akamine , Masahiro Oshikiri
IPC分类号： G10L19/08 , G06T9/00 , G10L19/00 , G10L19/12 , H03M7/30 , G10L7/02
CPC分类号： H03M7/3082 , G06T9/008
摘要： A vector quantizing apparatus includes a first search section for obtaining an approximate vector X1 which is approximated to a desired vector R, a residual vector calculator for calculating a residual vector Rv from the desired vector R and the approximate vector X1, a weighting section for obtaining weighted vectors X2 to XN of code vectors x2 to xN, and a second search section for calculating an estimation value which is the magnitude of a projection vector of the residual vector Rv with respect to the vector space formed by the approximate vector X1 and the weighted vectors X2 to XN, and searching a code vector which maximizes this estimation value.
摘要翻译：矢量量化装置包括用于获得近似于期望矢量R的近似矢量X1的第一搜索部分，用于从期望矢量R和近似矢量X1计算残差矢量Rv的残差矢量计算器，用于获得的加权部分以及第二搜索部分，用于计算相对于由近似矢量X1形成的向量空间的残差矢量Rv的投影矢量的大小的估计值，以及加权矢量X 2的加权矢量向量X2到XN，并且搜索最大化该估计值的码矢量。

7. 发明授权

US07191120B2 Speech encoding method, apparatus and program 失效
标题翻译：语音编码方法，装置和程序
公开(公告)号：US07191120B2
公开(公告)日：2007-03-13
申请号：US10675947
申请日：2003-10-02
申请人： Masahiro Oshikiri , Kimio Miseki , Masami Akamine
发明人： Masahiro Oshikiri , Kimio Miseki , Masami Akamine
IPC分类号： G10L19/00 , G10L11/04
CPC分类号： G10L25/93 , G10L19/09 , G10L25/78
摘要： A speech encoding method, apparatus and program wherein an input speech signal is divided into a plurality of frames each having a predetermined length, each of the frames is subdivided into a plurality of subframes, a predictive pitch period of a subframe in a to-be-encoded current frame is obtained by using pitch periods of at least two frames of the current frame and past and future frames with respect to the current frame; a pitch period of a subframe in the current frame is obtained by using the predictive pitch period, a relative pitch pattern codebook storing a plurality of relative pitch patterns representing fluctuations in pitch periods of a plurality of subframes is prepared, and a change in pitch period of plural subframes is expressed with one relative pitch pattern selected from the relative pitch pattern codebook.
摘要翻译：一种语音编码方法，装置和程序，其中输入语音信号被划分为具有预定长度的多个帧，每个帧被细分为多个子帧，子帧的预测音调周期通过使用当前帧的至少两帧的音调周期和相对于当前帧的过去和未来帧来获得编码的当前帧; 通过使用预测音调周期来获得当前帧中的子帧的音调周期，准备存储表示多个子帧的音调周期的波动的多个相对音调模式的相对音调模式码本，并且音调周期的变化由相对的音调图案码本中选择的一个相对音调图形来表示多个子帧。

8. 发明授权

US06427135B1 Method for encoding speech wherein pitch periods are changed based upon input speech signal 失效
标题翻译：用于编码语音的方法，其中音调周期根据输入的语音信号而改变
公开(公告)号：US06427135B1
公开(公告)日：2002-07-30
申请号：US09696962
申请日：2000-10-27
申请人： Kimio Miseki , Masahiro Oshikiri , Tadashi Amada , Masami Akamine
发明人： Kimio Miseki , Masahiro Oshikiri , Tadashi Amada , Masami Akamine
IPC分类号： G10L1104
CPC分类号： G10L19/012 , G10L19/02
摘要： A method for encoding speech wherein an input speech signal is separated by a component separator into a first component mainly constituted by speech and a second component mainly constituted by a background noise at each predetermined unit of time, a bit allocation selector selects bit allocation for each component based on the first and second components from among a plurality of predetermined candidates for bit allocation, a speech encoder and a noise encoder encode the first and second components from the component separator based on the bit allocation according to predetermined different methods for encoding, and a multiplexer multiplexes encoded data of the first and second components and information on the bit allocation and outputs them as transmitted encoded data.
摘要翻译：一种编码语音的方法，其中输入语音信号由分量分离器分离成主要由语音构成的第一分量和主要由每个预定单位时间的背景噪声构成的第二分量，比特分配选择器选择每个基于来自用于比特分配的多个预定候选中的第一和第二分量的分量，语音编码器和噪声编码器根据预定的不同编码方法基于比特分配对来自分量分离器的第一和第二分量进行编码，以及复用器复用第一和第二分量的编码数据和关于比特分配的信息，并将其作为发送的编码数据输出。

9. 发明授权

US6064962A Formant emphasis method and formant emphasis filter device 失效
标题翻译：渐进强调方法和共振峰强调过滤装置
公开(公告)号：US6064962A
公开(公告)日：2000-05-16
申请号：US713356
申请日：1996-09-13
申请人： Masahiro Oshikiri , Masami Akamine , Kimio Miseki , Akinobu Yamashita
发明人： Masahiro Oshikiri , Masami Akamine , Kimio Miseki , Akinobu Yamashita
IPC分类号： G10L19/26 , G10L25/15 , G10L5/02
CPC分类号： G10L19/26 , G10L25/15
摘要： In a formant emphasis method of emphasizing the formant as the spectral peak of an input speech signal and attenuating the spectral valley of the input speech signal, a spectrum emphasis filter performs processing for emphasizing the formant of the input speech signal and attenuating the valley of the input speech signal. A first-order variable characteristic filter whose characteristic adaptively changes in accordance with the characteristic of the input speech signal and a first-order fixed characteristic filter compensate a spectral tilt included in an output signal from the spectrum emphasis filter.
摘要翻译：在强调共振峰作为输入语音信号的频谱峰值并衰减输入语音信号的频谱谷的共振峰强调方法中，频谱加重滤波器执行用于强调输入语音信号的共振峰并衰减输入语音信号的谷值的处理输入语音信号。其特征根据输入语音信号的特性自适应地改变的一阶可变特征滤波器和一阶固定特性滤波器对包括在来自频谱加重滤波器的输出信号中的频谱倾斜进行补偿。

10. 发明授权

US5878387A Coding apparatus having adaptive coding at different bit rates and pitch emphasis 失效
标题翻译：编码装置具有不同比特率的自适应编码和音调强调
公开(公告)号：US5878387A
公开(公告)日：1999-03-02
申请号：US537276
申请日：1995-09-29
申请人： Masahiro Oshikiri , Kimio Miseki , Masami Akamine , Tadashi Amada
发明人： Masahiro Oshikiri , Kimio Miseki , Masami Akamine , Tadashi Amada
IPC分类号： G10L19/00 , G10L11/04 , G10L19/06 , H03M7/30 , H04B14/00 , G10L5/04
CPC分类号： G10L19/005 , G10L19/06 , G10L19/09
摘要： The coding apparatus comprises an adaptive codebook storing excitation signals as vectors, a synthesis filter for forming a synthesis signal, referring to the vectors stored in the adaptive codebook, a similarity computation circuit for computing a similarity between the synthesis signal obtained by the synthesis filter and a target signal, and a coding scheme determining circuit for deciding one coding scheme from a plurality of coding schemes respectively having coding bit rates different from each other, on the basis of the similarity obtained by the similarity computation circuit.
摘要翻译：编码装置包括存储作为向量的激励信号的自适应码本，用于形成合成信号的合成滤波器，参考存储在自适应码本中的矢量，相似度计算电路，用于计算由合成滤波器获得的合成信号与目标信号和编码方案决定电路，用于根据由相似度计算电路获得的相似度，从分别具有彼此不同的编码比特率的多个编码方案中确定一个编码方案。

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