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1. 发明授权

US5230036A Speech coding system utilizing a recursive computation technique for improvement in processing speed 失效
标题翻译：语音编码系统利用一种改进加工速度的计算技术
公开(公告)号：US5230036A
公开(公告)日：1993-07-20
申请号：US598989
申请日：1990-10-17
申请人： Masami Akamine , Yuji Okuda , Kimio Miseki
发明人： Masami Akamine , Yuji Okuda , Kimio Miseki
IPC分类号： G10L19/00 , G10L19/083 , G10L19/12 , G10L19/16
CPC分类号： G10L19/083 , G10L19/00 , G10L19/12 , G10L19/16
摘要： This invention provides a novel speech coding system which recursively executes a filter-applied "Toeplitz characteristic" by causing a drive signal (i.e., an excitation signal) to be converted into a "Toeplitz matrix" when detecting a pitch period in which distortion of the input vector and the vector subsequent to the application of filter-applied computation to the drive signal vector in the pitch forecast called either "closed loop" or "compatible code book" is minimized. The vector quantization method substantially making up the speech coding system of the invention is characteristically used by the system.

2. 再颁专利

USRE36646E Speech coding system utilizing a recursive computation technique for improvement in processing speed 失效
标题翻译：语音编码系统利用递归计算技术提高处理速度
公开(公告)号：USRE36646E
公开(公告)日：2000-04-04
申请号：US504227
申请日：1995-07-19
申请人： Masami Akamine , Yuji Okuda , Kimio Miseki
发明人： Masami Akamine , Yuji Okuda , Kimio Miseki
IPC分类号： G10L19/00 , G10L19/083 , G10L19/12 , G10L19/16 , G10L9/00
CPC分类号： G10L19/083 , G10L19/00 , G10L19/12 , G10L19/16
摘要： This invention provides a novel speech coding system which recursively executes a filter-applied "Toeplitz characteristic" by causing a drive signal (i.e., an excitation signal) to be converted into a "Toeplitz matrix" when detecting a pitch period in which distortion of the input vector and the vector subsequent to the application of filter-applied computation to the drive signal vector in the pitch forecast called either "closed loop" or "compatible code book" is minimized. The vector quantization method substantially making up the speech coding system of the invention is characteristically used by the system.
摘要翻译：本发明提供了一种新颖的语音编码系统，其在检测到音调周期时，通过使驱动信号（即，激励信号）转换成“Toeplitz矩阵”，递归地执行滤波器施加的“Toeplitz特性” 输入向量和将滤波器应用的计算应用于在称为“闭环”或“兼容码本”的音调预测中的驱动信号矢量之后的矢量被最小化。基本上构成本发明的语音编码系统的矢量量化方法由系统特征性地使用。

3. 发明授权

US5150387A Variable rate encoding and communicating apparatus 失效
标题翻译：可变速率编码和通信装置
公开(公告)号：US5150387A
公开(公告)日：1992-09-22
申请号：US630911
申请日：1990-12-20
申请人： Hidetaka Yoshikawa , Kimio Miseki , Masami Akamine
发明人： Hidetaka Yoshikawa , Kimio Miseki , Masami Akamine
IPC分类号： H04B1/66
CPC分类号： H04B1/667 , G10L19/24
摘要： In a transmitter in the present invention, an input signal is input to a QMF bank 102 where the input signal is divided to a plurality of frequency bands to form corresponding band signals. A distributed bit calculating unit 109 calculates respective bit rates with which the corresponding band signals are encoded on the respective power values of the band signals. Quantizers 104-1, 104-2, . . . , 104-n encode the respective band signals at the corresponding bit rates and input the resulting corresponding band codes to a multiplexer unit 111 which incorporates the respective band codes into a cell as an information unit and sends the cell. In a receiver, a cell is decomposed to obtain the respective band codes, which are then dequantized to form the corresponding band signals. These band signals are synthesized to form a signal for the entire band, and the signal for the entire band is output as a decoded signal.
摘要翻译：在本发明的发射机中，将输入信号输入到QMF组102，其中输入信号被划分成多个频带以形成相应的频带信号。分布位计算单元109计算相应频带信号对频带信号的各个功率值进行编码的各个比特率。量子化器104-1,104-2，。。，104-n以对应的比特率对各个频带信号进行编码，并将所得到的相应频带码输入到将各频带码合并到一个小区中作为信息单元的多路复用单元111，并发送该小区。在接收机中，单元被分解以获得相应的频带码，然后将它们去量化以形成相应的频带信号。这些频带信号被合成以形成整个频带的信号，并且将整个频带的信号作为解码信号输出。

4. 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。

5. 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。

6. 再颁专利

USRE36721E Speech coding and decoding apparatus 失效
标题翻译：语音编解码装置
公开(公告)号：USRE36721E
公开(公告)日：2000-05-30
申请号：US561751
申请日：1995-11-22
申请人： Masami Akamine , Kimio Miseki
发明人： Masami Akamine , Kimio Miseki
IPC分类号： G10L9/00
摘要： A speech signal is input to an excitation signal generating section, a prediction filter and a prediction parameter calculator. The prediction parameter calculator calculates a predetermined number of prediction parameters (LPC parameter or reflection coefficient) by an autocorrelation method or covariance method, and supplies the acquired prediction parameters to a prediction parameter coder. The codes of the prediction parameters are sent to a decoder and a multiplexer. The decoder sends decoded values of the codes of the prediction parameters to the prediction filter and the excitation signal generating section. The prediction filter calculates a prediction residual signal, which is the difference between the input speech signal and the decoded prediction parameter, and sends it to the excitation signal generating section. The excitation signal generating section calculates the pulse interval and amplitude for each of a predetermined number of subframes based on the input speech signal, the prediction residual signal and the quantized value of the prediction parameter, and sends them to the multiplexer. The multiplexer combines these codes and the codes of the prediction parameters, and send the results as an output signal of a coding apparatus to a transmission path or the like.
摘要翻译：语音信号被输入到激励信号产生部分，预测滤波器和预测参数计算器。预测参数计算器通过自相关方法或协方差方法计算预定数量的预测参数（LPC参数或反射系数），并将所获取的预测参数提供给预测参数编码器。预测参数的代码被发送到解码器和多路复用器。解码器将预测参数的代码的解码值发送到预测滤波器和激励信号生成部。预测滤波器计算作为输入语音信号和解码预测参数之间的差的预测残差信号，并将其发送到激励信号生成部。激励信号生成部基于输入的语音信号，预测残差信号和预测参数的量化值，计算预定数量的子帧中的每一个的脉冲间隔和幅度，并将其发送到多路复用器。多路复用器组合这些代码和预测参数的代码，并将结果作为编码装置的输出信号发送到传输路径等。

7. 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。

8. 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。

9. 发明授权

US5265167A Speech coding and decoding apparatus 失效
标题翻译：语音编解码装置
公开(公告)号：US5265167A
公开(公告)日：1993-11-23
申请号：US13551
申请日：1992-11-19
申请人： Masami Akamine , Kimio Miseki
发明人： Masami Akamine , Kimio Miseki
IPC分类号： G10L19/04 , G10L19/10 , G10L9/00
CPC分类号： G10L19/113
摘要： A speech signal is input to an excitation signal generating section, a prediction filter and a prediction parameter calculator. The prediction parameter calculator calculates a predetermined number of prediction parameters (LPC parameter or reflection coefficient) by an autocorrelation method or covariance method, and supplies the acquired prediction parameters to a prediction parameter coder. The codes of the prediction parameters are sent to a decoder and a multiplexer. The decoder sends decoded values of the codes of the prediction parameters to the prediction filter and the excitation signal generating section. The prediction filter calculates a prediction residual signal, which is the difference between the input speech signal and the decoded prediction parameter, and sends it to the excitation signal generating section. The excitation signal generating section calculates the pulse interval and amplitude for each of a predetermined number of subframes based on the input speech signal, the prediction residual signal and the quantized value of the prediction parameter, and sends them to the multiplexer. The multiplexer combines these codes and the codes of the prediction parameters, and send the results as an output signal of a coding apparatus to a transmission path or the like.
摘要翻译：语音信号被输入到激励信号产生部分，预测滤波器和预测参数计算器。预测参数计算器通过自相关方法或协方差方法计算预定数量的预测参数（LPC参数或反射系数），并将所获取的预测参数提供给预测参数编码器。预测参数的代码被发送到解码器和多路复用器。解码器将预测参数的代码的解码值发送到预测滤波器和激励信号生成部。预测滤波器计算作为输入语音信号和解码预测参数之间的差的预测残差信号，并将其发送到激励信号生成部。激励信号生成部基于输入的语音信号，预测残差信号和预测参数的量化值，计算预定数量的子帧中的每一个的脉冲间隔和幅度，并将其发送到多路复用器。多路复用器组合这些代码和预测参数的代码，并将结果作为编码装置的输出信号发送到传输路径等。

10. 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。

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