公开(公告)号：US08249866B2

公开(公告)日：2012-08-21

申请号：US12751191

申请日：2010-03-31

申请人： Kimio Miseki

发明人： Kimio Miseki

IPC分类号： G10L19/08 ,  G10L19/12

CPC分类号： G10L19/18

摘要： A speech decoding method which generates an excitation signal and a synthesis filter from coded data and which obtains a speech signal based on the excitation signal and the synthesis filter. The method includes acquiring identification information used for determining whether the speech signal to be decoded is a narrowband signal or a wideband signal; and modifying the excitation signal based on the identification information by controlling strength or presence of emphasis of pitch periodicity with respect to the excitation signal generated from the coded data, so as to generate the speech signal by use of the modified excitation signal and the synthesis filter.

摘要翻译： 一种语音解码方法，其从编码数据生成激励信号和合成滤波器，并且基于激励信号和合成滤波器获得语音信号。 该方法包括获取用于确定要解码的语音信号是窄带信号还是宽带信号的识别信息; 以及通过相对于从编码数据产生的激励信号控制音调周期的强度的强度或存在，基于识别信息修改激励信号，以便通过使用修改的激励信号和合成滤波器来产生语音信号 。

公开(公告)号：US20110206220A1

公开(公告)日：2011-08-25

申请号：US12949437

申请日：2010-11-18

申请人： Norikatsu Chiba ,  Kimio Miseki

发明人： Norikatsu Chiba ,  Kimio Miseki

IPC分类号： H03G3/00

CPC分类号： H04R1/1091 ,  G10L21/02 ,  G10L21/0232 ,  H03G5/005 ,  H03G9/005 ,  H04R1/1016 ,  H04R25/353 ,  H04R25/356 ,  H04R2460/15

摘要： According to one embodiment, an acoustic signal corrector includes: an output module; a selection receiver; and a holder. The output module is configured to output a plurality of acoustic signals. Amplitude values of frequencies within a frequency band of each of the acoustic signals are emphasized as emphasized amplitude values. A plurality of amplitude values among the emphasized amplitude values are corrected as corrected amplitude values at some of frequencies within the frequency band. Resonance is possibly induced in the frequency band by sealing an ear canal. The selection receiver is configured to receive a selection of one of the acoustic signals output by the output module. The holder is configured to hold, as a configuration for sound quality correction, a configuration corresponding to the correction of the one of the acoustic signals at the some of the frequencies.

摘要翻译： 根据一个实施例，声信号校正器包括：输出模块; 选择接收机; 和持有人。 输出模块被配置为输出多个声信号。 每个声信号的频带内的频率的振幅值被强调为强调的振幅值。 强调幅度值中的多个振幅值被校正为在频带内的一些频率处的校正振幅值。 通过密封耳道可能在频带中引起共振。 选择接收器被配置为接收由输出模块输出的声信号之一的选择。 保持器被配置为作为用于声音质量校正的配置来保持与在一些频率处的声信号之一的校正相对应的配置。

公开(公告)号：US07698296B2

公开(公告)日：2010-04-13

申请号：US11699814

申请日：2007-01-30

申请人： Chikashi Sugiura ,  Takehiko Isaka ,  Kimio Miseki

发明人： Chikashi Sugiura ,  Takehiko Isaka ,  Kimio Miseki

IPC分类号： G06F7/00

CPC分类号： G06F17/30029 ,  G06F17/30053 ,  Y10S707/99943

摘要： A content reproducing apparatus includes a display unit configured to display a play list and candidate contents able to be added to the play list, a selection unit configured to select, from the candidate contents, an undesired content which a user does not want to add to the play list, a calculation unit configured to calculate a first retrieval statistical quantity based on first characteristic quantity of the undesired content or a second retrieval statistical quantity based on second characteristic quantity of a desired content which the user wants to add to the play list, and a retrieve unit configured to retrieve the candidate contents to prepare the play list, in accordance with similarity which has been calculated by using the first or second retrieval statistical quantity and which shows to which a given content having third characteristic quantity is similar, the desired content or the undesired content.

摘要翻译： 内容再现装置包括：显示单元，被配置为显示播放列表和能够添加到播放列表的候选内容;选择单元，被配置为从候选内容中选择用户不想添加的不需要的内容 播放列表，计算单元，被配置为基于用户想要添加到播放列表的期望内容的第二特征量，基于不需要内容的第一特征量或第二检索统计量来计算第一检索统计量， 以及检索单元，其被配置为根据已经通过使用第一或第二检索统计量计算并且示出具有第三特征量的给定内容相似的相似度来检索候选内容以准备播放列表，期望的 内容或不需要的内容。

公开(公告)号：US20100057476A1

公开(公告)日：2010-03-04

申请号：US12549659

申请日：2009-08-28

申请人： Takashi SUDO ,  Kimio Miseki

发明人： Takashi SUDO ,  Kimio Miseki

IPC分类号： G10L19/00

CPC分类号： G10L21/038

摘要： A signal bandwidth extension apparatus includes a determination unit which determines whether or not a peak component of the input signal is lacked in the band to be extended, and a control unit which controls to extend the bandwidth when the determination unit determines that the peak component of the input signal is lacked in the band to be extended, and not to extend the bandwidth when the determination unit determines that the peak component is not lacked.

摘要翻译： 信号带宽扩展装置包括确定单元，其确定在要扩展的频带中输入信号的峰值分量是否不足;以及控制单元，其在所述确定单元确定所述峰值分量 在要延长的频带中缺少输入信号，并且当确定单元确定不存在峰值分量时不延长带宽。

公开(公告)号：US20080247557A1

公开(公告)日：2008-10-09

申请号：US12045457

申请日：2008-03-10

申请人： Takashi Sudo ,  Kimio Miseki ,  Yuji Kawashima

发明人： Takashi Sudo ,  Kimio Miseki ,  Yuji Kawashima

IPC分类号： H04B3/20

CPC分类号： H04M9/082

摘要： According to one embodiment, a signal processing apparatus includes a speaker configured to output the received input signal on which a delay detection signal which has a frequency component of an inaudible frequency on a received input signal is superposed to an acoustic space, an extracting section configured to extract the delay detection signal from the sending input signal outputted from microphone configured to collect sound in the acoustic space a calculating section configured to calculate a delay time between the received input signal and an acoustic echo component contained in the sending input signal, a delay section configured to delay the received input signal by a time corresponding to the delay time and generate a delayed received input signal, and an echo suppression processing section configured to suppress the acoustic echo component contained in the sending input signal by use of the delayed received input signal.

摘要翻译： 根据一个实施例，一种信号处理装置包括：扬声器，被配置为输出接收到的输入信号，在所述接收输入信号上将具有接收到的输入信号的不可听频率的频率分量的延迟检测信号叠加到声学空间;提取部分配置 从被配置为在声学空间中收集声音的麦克风输出的发送输入信号中提取延迟检测信号，计算部分，被配置为计算接收的输入信号和包含在发送输入信号中的声学回声分量之间的延迟时间，延迟 被配置为将接收的输入信号延迟与延迟时间相对应的时间，并产生延迟的接收输入信号;以及回波抑制处理部，被配置为通过使用延迟的接收输入来抑制包含在发送输入信号中的声学回声分量 信号。

公开(公告)号：US20060020450A1

公开(公告)日：2006-01-26

申请号：US11240495

申请日：2005-10-03

申请人： Kimio Miseki

发明人： Kimio Miseki

IPC分类号： G10L19/10

CPC分类号： G10L19/18

摘要： A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.

摘要翻译： 一种宽带语音编码方法，包括识别输入语音信号是窄带信号还是宽带信号，以及通过基于识别结果控制宽带语音编码处理的预定参数来对输入语音信号进行编码。

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。

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。

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。

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。