专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

51. 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。

52. 发明授权

US06842732B2 Speech encoding and decoding method and electronic apparatus for synthesizing speech signals using excitation signals 有权
标题翻译：语音编码和解码方法以及使用激励信号合成语音信号的电子装置
公开(公告)号：US06842732B2
公开(公告)日：2005-01-11
申请号：US09803998
申请日：2001-03-13
申请人： Kimio Miseki
发明人： Kimio Miseki
IPC分类号： G10L19/12 , G10L19/00 , G10L19/06 , G10L19/08 , H03M7/30
CPC分类号： G10L19/08 , G10L2019/0011
摘要： A speech encoding method of generating a synthesized speech signal by using an excitation signal generated by using an adaptive codebook storing a past excitation signal includes the steps of modifying an excitation signal used to generate a synthesized speech signal by filter processing, and storing the modified excitation signal in the adaptive codebook.
摘要翻译：通过使用通过使用存储过去激励信号的自适应码本生成的激励信号来生成合成语音信号的语音编码方法包括以下步骤：通过滤波处理修改用于生成合成语音信号的激励信号，并存储修改的激励信号在自适应码本中。

53. 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。

54. 发明授权

US6131083A Method of encoding and decoding speech using modified logarithmic transformation with offset of line spectral frequency 失效
标题翻译：使用线谱频率偏移的改进对数变换对语音进行编码和解码的方法
公开(公告)号：US6131083A
公开(公告)日：2000-10-10
申请号：US219773
申请日：1998-12-23
申请人： Kimio Miseki , Katsumi Tsuchiya
发明人： Kimio Miseki , Katsumi Tsuchiya
IPC分类号： G10L13/00 , G10L19/00 , G10L19/04 , G10L19/06 , H03M7/30 , H04B14/04
CPC分类号： G10L19/07
摘要： On the basis of an autocorrelation coefficient calculated by an autocorrelation coefficient computation section from an input speech signal, an LSF computation section computes LSF parameters F(k) (k=1, 2, . . . , N). A modified logarithmic transformation section performs on the LSF parameters a logarithmic transformation with offset defined by f(k)=logC (1+A.times.F(k)) to obtain modified logarithmic LSF parameters f(k). The resulting modified logarithmic LSF parameters are quantized by a quantization section to provide quantized LSF parameters fq(k). Codes representing the quantized LSF parameters fq(k) are outputted. An inverse transformation defined by Fq(k)=(C.sup.fq(k) -1)/A is performed on the LSF parameters fq(k) to output LSF parameters Fq(k) on the general frequency scale.
摘要翻译：基于自相关系数计算部分从输入语音信号计算的自相关系数，LSF计算部分计算LSF参数F（k）（k = 1,2，...，N）。改进的对数变换部分对LSF参数执行由f（k）= logC（1 + AxF（k））定义的偏移的对数变换，以获得修正的对数LSF参数f（k）。所得到的修正对数LSF参数由量化部分量化，以提供量化的LSF参数fq（k）。输出表示量化的LSF参数fq（k）的代码。对LSF参数fq（k）执行由Fq（k）=（Cfq（k）-1）/ A定义的逆变换，以在一般频率标度上输出LSF参数Fq（k）。

55. 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。

56. 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。

57. 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。

58. 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。

59. 发明授权

US5671327A Speech encoding apparatus utilizing stored code data 失效
标题翻译：利用存储码数据的语音编码装置
公开(公告)号：US5671327A
公开(公告)日：1997-09-23
申请号：US7710
申请日：1993-01-22
申请人： Masami Akamine , Masahiro Oshikiri , Kimio Miseki
发明人： Masami Akamine , Masahiro Oshikiri , Kimio Miseki
IPC分类号： G10L19/00 , G10L19/12 , G10L3/02 , G10L9/00
CPC分类号： G10L19/12
摘要： A learning-type speech encoding apparatus comprises an adaptive code book storing driving signal vectors, a minimum distortion searching circuit for searching the adaptive code book for an optimum driving signal vector on the basis of the input speech signal, a synthesizing filter for synthesizing a speech signal using the optimum driving signal vector retrieved, a buffer for storing the optimum driving signal vector retrieved, a training vector creating section for producing a training vector by segmenting the stored driving signal vector in units of a specified length, and a learning section for learning by constantly updating the driving signal vectors in the code book on the basis of the training vector.
摘要翻译：一种学习型语音编码装置，包括存储驱动信号矢量的自适应码本，用于基于输入的语音信号搜索自适应码本以获得最佳驱动信号向量的最小失真搜索电路，用于合成语音的合成滤波器使用所检索的最佳驱动信号矢量的信号，用于存储所检索的最佳驱动信号矢量的缓冲器，用于通过以指定长度为单位分割存储的驾驶信号向量来产生训练矢量的训练矢量创建部分，以及用于学习的学习部分通过基于训练矢量不断更新码本中的驾驶信号矢量。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式