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1. 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。

2. 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。

3. 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。

4. 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。

5. 发明申请

US20130028429A1 INFORMATION PROCESSING APPARATUS AND METHOD OF PROCESSING AUDIO SIGNAL FOR INFORMATION PROCESSING APPARATUS 审中-公开
标题翻译：信息处理装置和处理用于信息处理装置的音频信号的方法
公开(公告)号：US20130028429A1
公开(公告)日：2013-01-31
申请号：US13462163
申请日：2012-05-02
申请人： Tadashi Amada , Toshifumi YAMAMOTO , Kimio MISEKI
发明人： Tadashi Amada , Toshifumi YAMAMOTO , Kimio MISEKI
IPC分类号： H04R29/00
CPC分类号： H04R3/04 , H04R1/1016 , H04R1/1091 , H04R29/001
摘要： According to one embodiment, an information processing apparatus includes a measurement module and a correction filter design module. The measurement module measures frequency characteristics of an earphone connected to an output terminal using a measurement audio signal output from the output terminal and collected by a microphone. The correction filter design module designs a correction filter in association with one range of a treble range higher than a crossover frequency range and a bass range lower than the crossover frequency range, based on the measured frequency characteristics of the earphone and predetermined goal frequency characteristics.
摘要翻译：根据一个实施例，信息处理设备包括测量模块和校正滤波器设计模块。测量模块使用从输出端子输出并由麦克风收集的测量音频信号来测量连接到输出端子的耳机的频率特性。校正滤波器设计模块基于测量的耳机的频率特性和预定的目标频率特性，设计校正滤波器，该校正滤波器与高于交越频率范围的高音范围和低于交越频率范围的低音范围相关联。

6. 发明授权

US06385576B2 Speech encoding/decoding method using reduced subframe pulse positions having density related to pitch 失效
标题翻译：使用具有与音调相关的密度的减少的子帧脉冲位置的语音编码/解码方法
公开(公告)号：US06385576B2
公开(公告)日：2002-05-07
申请号：US09220062
申请日：1998-12-23
申请人： Tadashi Amada , Kimio Miseki
发明人： Tadashi Amada , Kimio Miseki
IPC分类号： G10L1908
CPC分类号： G10L19/10
摘要： A speech encoding method in which information representing characteristics of a synthesis filter is generated based on an input speech signal in units of one frame. A pitch vector is generated from an adaptive codebook containing past excitation signals, and a first number of reduced pulse position candidates are generated by selecting a first number of pulse positions from a number of possible pulse positions in each of sub-frames obtained by dividing the frame, where a density of the reduced pulse position candidates is high where the pitch vector has a large power and decreases in accordance with a decrease in the power. A second number of pulse positions is selected from the reduced pulse position candidates to generate a pulse train having a plurality of pulses located at pulse positions corresponding to a second number of pulse positions under the criterion of minimizing an error between the input speech signal and a synthesis signal which is an output of the synthesis filter whose input is an excitation signal generated by adding the pitch vector and the pulse train.
摘要翻译：一种语音编码方法，其中基于一帧的单位的输入语音信号来生成表示合成滤波器的特性的信息。从包含过去的激励信号的自适应码本生成音调矢量，并且通过从通过划分所得到的每个子帧中的每个子帧中的多个可能的脉冲位置中选择第一数量的脉冲位置来生成第一数量的缩小脉冲位置候选帧，其中缩小脉冲位置候选的密度高，其中音调矢量具有大的功率，并且根据功率的降低而减小。从缩小的脉冲位置候选中选择第二数量的脉冲位置，以产生脉冲序列，该脉冲串具有位于与第二数量的脉冲位置相对应的脉冲位置的多个脉冲，该标准在最小化输入语音信号与a 合成信号是合成滤波器的输出，其输入是通过将音调矢量和脉冲串相加而产生的激励信号。

7. 发明申请

US20130223636A1 MEASUREMENT APPARATUS AND MEASUREMENT METHOD 审中-公开
标题翻译：测量装置和测量方法
公开(公告)号：US20130223636A1
公开(公告)日：2013-08-29
申请号：US13671963
申请日：2012-11-08
申请人： Tadashi Amada , Toshifumi Yamamoto , Kimio Miseki
发明人： Tadashi Amada , Toshifumi Yamamoto , Kimio Miseki
IPC分类号： H04R29/00
CPC分类号： H04R29/001 , H04R1/1016 , H04R2430/01 , H04R2430/03
摘要： According to one embodiment, a measurement apparatus is configured to supply, to a output device, a second measurement signal to sweep the frequency, to receive a second received signal output from the output device when the second measurement signal is supplied to the output device, and to calculate an impulse response by convolving the second received signal and an inverted measurement signal having an inverted characteristic of the second measurement signal. The second measurement signal has a signal characteristic obtained by multiplying the first measurement signal by a characteristic of a characteristic of the amplitude spectrum.
摘要翻译：根据一个实施例，一种测量装置被配置为当输出第二测量信号被提供给输出装置时，向输出装置提供第二测量信号以扫描频率，以接收从输出装置输出的第二接收信号，并且通过卷积第二接收信号和具有第二测量信号的反相特性的反相测量信号来计算脉冲响应。第二测量信号具有通过将第一测量信号乘以振幅谱的特性的特性而获得的信号特性。

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

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。

10. 发明授权

US5774838A Speech coding system utilizing vector quantization capable of minimizing quality degradation caused by transmission code error 失效
标题翻译：利用能够最小化由传输码错误引起的质量劣化的矢量量化的语音编码系统
公开(公告)号：US5774838A
公开(公告)日：1998-06-30
申请号：US536362
申请日：1995-09-29
申请人： Kimio Miseki , Tadashi Amada
发明人： Kimio Miseki , Tadashi Amada
IPC分类号： G10L19/005 , G10L3/00
CPC分类号： G10L19/005
摘要： In a vector quantization apparatus for expressing a target vector by using a code vector designated by an index, an error evaluating section performs error evaluation for a code vector without considering a code error of the index and error evaluation with considering the code error, a first selecting section selects a small number of indexes from a larger number of indexes on the basis of an evaluation result without considering the code error, and a second selecting section selects, on the basis of an evaluation result with considering the code error, an index used to express the target vector from a small number of indexes selected by the first selecting section.
摘要翻译：在通过使用由索引指定的代码矢量来表示目标矢量的矢量量化装置中，误差评估部分对代码矢量执行错误评估，而不考虑代码误差的索引和错误评估的代码错误，第一选择部分基于评估结果从更大数量的索引中选择少量的索引，而不考虑代码错误，并且第二选择部分基于考虑到代码错误的评估结果来选择使用的索引以从由第一选择部分选择的少量索引来表达目标向量。

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