公开(公告)号：JP2010217912A

公开(公告)日：2010-09-30

申请号：JP2010112494

申请日：2010-05-14

Applicant: Motorola Inc ,  モトローラ・インコーポレイテッドMotorola Incorporated

Inventor： JASIUK MARK A ,  RAMABADRAN TENKASI V ,  MITTAL UDAR ,  ASHLEY JAMES P ,  MCLAUGHLIN MICHAEL J

IPC: G10L19/08 ,  G10L19/14 ,  H03M7/30

CPC classification number: G10L19/09

Abstract: PROBLEM TO BE SOLVED: To provide a method and apparatus for performing speech coding which improves speech quality. SOLUTION: The method for encoding speech by a speech coding device comprises: a stage for receiving an input signal; a stage for creating an object vector on the basis of the input signal; a stage for creating a plurality of weighted adaptive code book vectors on the basis of a signal sub-sample resolution delay value, an adaptive code book, and a weighted synthesis filter; a stage for creating a weighted fixed code book(FCB) exiting vector on the basis of the object vector and the plurality of weighted adaptive code book vector; a stage for creating a plurality of correlation terms on the basis of the object vector, the plurality of weighted adaptive code book vectors and the weighted FCB exciting vector; and a stage for selecting a gain vector from a table in response to an error minimizing reference. The gain vector comprises at least two adaptive code book gains and one fixed code book gain, and the error minimizing reference is based on the plurality of correlation terms. COPYRIGHT: (C)2010,JPO&INPIT

Abstract translation: 要解决的问题：提供一种提高语音质量的用于执行语音编码的方法和装置。 解决方案：由语音编码装置编码语音的方法包括：用于接收输入信号的级; 基于输入信号创建对象向量的阶段; 基于信号子采样分辨率延迟值，自适应码本和加权合成滤波器创建多个加权自适应码本矢量的阶段; 基于所述对象矢量和所述多个加权自适应编码矢量生成退出向量的加权固定码本（FCB）的阶段; 基于对象矢量，多个加权自适应码本矢量和加权FCB激励矢量创建多个相关项的阶段; 以及用于响应于误差最小化参考从表中选择增益矢量的阶段。 增益矢量包括至少两个自适应码本增益和一个固定码本增益，并且误差最小化参考是基于多个相关项。 版权所有（C）2010，JPO＆INPIT

公开(公告)号：JP2000155597A

公开(公告)日：2000-06-06

申请号：JP35934599

申请日：1999-12-17

Applicant: MOTOROLA INC

Inventor： GERSON IRA A ,  JASIUK MARK A ,  HARTMAN MATTHEW A

IPC: G10L19/12 ,  H03H17/00 ,  H03H17/02 ,  H03M7/30 ,  H03M7/38 ,  H04B14/04 ,  G10L19/00 ,  G10L19/06

Abstract: PROBLEM TO BE SOLVED: To provide a digital voice coding method improving controlls of filters of a voice encoder without increasing the complexity of the voice encoder accordingly. SOLUTION: A single filter providing controls of plural filters without complexity of the plural filters is realized by using an R-order filter for modeling frequency responses of the plural filters. The R-order filter can be used as a spectral noise weight filter (132) or the combination of the short term predictive encoder filter called as a synthesis filter in which noise are spectrally weighted and the spectral noise weighting filter in accordance with which embodiments are to be used. Generally, this method performs the moldeling of frequency responses of L pieces of P-order filters with the single R-order filter and, in this case, the order of R is made to be

公开(公告)号：WO9941736A3

公开(公告)日：1999-10-21

申请号：PCT/US9902431

申请日：1999-02-04

Applicant: MOTOROLA INC

Inventor： ASHLEY JAMES PATRICK ,  PENG WEIMIN ,  JASIUK MARK A ,  SMITH AARON M

IPC: G10L19/04 ,  G10L20060101 ,  G10L19/06 ,  G10L19/12 ,  H03M7/30 ,  G10L3/00 ,  G10L9/14

CPC classification number: G10L19/04 ,  G10L19/06 ,  G10L19/12 ,  H03M7/30

Abstract: A method and system for providing split vector quantization for use in determining constrained ordered set values, such as line spectrum pair parameters to determine spectral parameters in a data compression system, utilizes multiple codebooks (22a-22c) containing delta coded constrained ordered set values that are normalized to an upper and lower bound. An LSP reconstructor (34) reconstructs received spectral parameters to decode data, such as speech, based on the normalized delta quantization data of line spectrum pair parameters obtained from the split vector reconstruction codebooks (22a-22c). The LSP reconstructor (34) dynamically generates line spectrum pair parameters based on the normalized delta quantization data. In another embodiment, instead of storing the absolute value of the line spectrum pair parameters in segmented codebooks, the combination of at least two absolute value vectors and at least one normalized delta quantization vector is used for spectral quantization.

Abstract translation: 一种用于提供用于确定约束有序集合值（例如用于确定数据压缩系统中的频谱参数的线谱对参数）的分割矢量量化的方法和系统利用包含增量编码约束有序集合值的多个码本（22a-22c） 被归一化为上限和下限。 基于从分割矢量重建码本（22a-22c）获得的线谱对参数的归一化的量化数据，LSP重构器（34）重建接收的频谱参数以对诸如语音的数据进行解码。 LSP重建器（34）基于归一化的量化数据动态地生成线路频谱对参数。 在另一个实施例中，代替在分段码本中存储线谱对参数的绝对值，而不是将至少两个绝对值向量和至少一个归一化的量化矢量的组合用于频谱量化。

公开(公告)号：EP1495465A4

公开(公告)日：2005-05-18

申请号：EP03745516

申请日：2003-02-14

Applicant: MOTOROLA INC

Inventor： RAMABADRAN TENKASI V ,  SMITH AARON M ,  JASIUK MARK A

IPC: G10L19/06 ,  G10L19/08 ,  G10L19/02 ,  G10L19/04

CPC classification number: G10L19/06 ,  G10L19/087

Abstract: A system or method for modeling a signal, such as a speech signal, wherein harmonic frequencies and amplitudes are identified (106) and the harmonic magnitudes are interpolated (110) to obtain spectral magnitudes at a set of fixed frequencies. An inverse transform is applied (112) to the spectral magnitudes to obtain a pseudo auto-correlation sequence, from which linear prediction coefficients are calculated (114). From the linear prediction coefficients, model harmonic magnitudes are generated by sampling the spectral envelope (118) defined by the linear prediction coefficients. A set of scale factors are then calculated (120) as the ratio of the harmonic magnitudes to the model harmonic magnitudes and interpolated to obtain a second set of scale factors (122) at the set of fixed frequencies. The spectral envelope magnitudes at the set of fixed frequencies (124) are multiplied by the second set of scale factors (126) to obtain new spectral magnitudes and the process is iterated to obtain final linear prediction coefficients.

公开(公告)号：WO2007047037A3

公开(公告)日：2009-04-09

申请号：PCT/US2006037408

申请日：2006-09-26

Applicant: MOTOROLA INC ,  JASIUK MARK A ,  RAMABADRAN TENKASI V

Inventor： JASIUK MARK A ,  RAMABADRAN TENKASI V

IPC: G10L19/04 ,  G10L19/08 ,  G10L19/12

CPC classification number: G10L19/26

Abstract: A speech communication system provides a speech encoder [100] that generates a set of coded parameters representative of the desired speech signal characteristics. The speech communication system also provides a speech decoder [200] that receives the set of coded parameters to generate reconstructed speech. The speech decoder includes an equalizer [204] that computes a matching set of parameters from the reconstructed speech [301] generated by the speech decoder [200], undoes the set of characteristics corresponding to the computed set of parameters, and imposes the set of characteristics corresponding to the coded set of parameters, thereby producing equalized reconstructed speech [306].

Abstract translation: 语音通信系统提供语音编码器[100]，其产生表示所需语音信号特性的编码参数集合。 语音通信系统还提供一种语音解码器[200]，其接收编码参数集合以产生重构语音。 语音解码器包括：均衡器[204]，其从语音解码器[200]生成的重构语音[301]中计算一组匹配的参数，撤销对应于所计算的参数集合的一组特征，并且将 特征对应于编码的参数集合，从而产生均衡的重构语音[306]。

公开(公告)号：WO2006007342A3

公开(公告)日：2006-03-02

申请号：PCT/US2005020299

申请日：2005-06-09

Applicant: MOTOROLA INC ,  KUSHNER WILLIAM M ,  HARTON SARA M ,  JASIUK MARK A

Inventor： KUSHNER WILLIAM M ,  HARTON SARA M ,  JASIUK MARK A

IPC: G10L15/00 ,  G10L15/06 ,  G10L21/00

CPC classification number: G10L17/26 ,  A62B18/08 ,  B63C11/26 ,  G10L15/20 ,  G10L21/0208

Abstract: A method for characterizing inhalation noise within a pressurized air delivery system, the method including the steps of: generating an inhalation noise model (912, 1012) based on inhalation noise; receiving an input signal (802) that includes inhalation noise comprising at least one inhalation noise burst; comparing (810) the input signal to the noise model to obtain a similarity measure; comparing the similarity measure to at least one threshold (832, 834) to detect the at least one inhalation noise burst; and characterizing (1354, 1356) the at least one detected inhalation noise burst.

Abstract translation: 一种用于表征加压空气输送系统内的吸入噪声的方法，所述方法包括以下步骤：基于吸入噪声产生吸入噪声模型（912,1012）; 接收包括至少一个吸入噪声突发的吸入噪声的输入信号（802）; 将输入信号与噪声模型进行比较（810）以获得相似性度量; 将所述相似性度量与至少一个阈值（832,834）进行比较以检测所述至少一个吸入噪声突发; 以及表征（1354,1356）所述至少一个检测到的吸入噪声突发。

公开(公告)号：WO2006007290B1

公开(公告)日：2006-08-24

申请号：PCT/US2005019827

申请日：2005-06-06

Applicant: WARE MOTOROLA INC A CORP OF TH ,  KUSHNER WILLIAM M ,  HARTON SARA M ,  JASIUK MARK A

Inventor： KUSHNER WILLIAM M ,  HARTON SARA M ,  JASIUK MARK A

IPC: G10L21/02 ,  G10L15/20

CPC classification number: B63C11/26 ,  G10L15/20 ,  G10L21/0208 ,  G10L25/78 ,  G10L2021/03643

Abstract: A method for equalizing a speech signal generated within a pressurized air delivery system, the method including the steps of: generating an inhalation noise model (1152) based on inhalation noise; receiving an input signal (802) that includes a speech signal; and equalizing the speech signal (1156) based on the noise model.

Abstract translation: 一种用于均衡在加压空气输送系统内产生的语音信号的方法，所述方法包括以下步骤：基于吸入噪声产生吸入噪声模型（1152）; 接收包括语音信号的输入信号（802）; 以及基于噪声模型来均衡语音信号（1156）。

公开(公告)号：WO2006007290A3

公开(公告)日：2006-06-01

申请号：PCT/US2005019827

申请日：2005-06-06

Applicant: MOTOROLA INC ,  KUSHNER WILLIAM M ,  HARTON SARA M ,  JASIUK MARK A

Inventor： KUSHNER WILLIAM M ,  HARTON SARA M ,  JASIUK MARK A

IPC: G10L21/02 ,  G10L15/20

CPC classification number: B63C11/26 ,  G10L15/20 ,  G10L21/0208 ,  G10L25/78 ,  G10L2021/03643

Abstract: A method for equalizing a speech signal generated within a pressurized air delivery system, the method including the steps of: generating an inhalation noise model (1152) based on inhalation noise; receiving an input signal (802) that includes a speech signal; and equalizing the speech signal (1156) based on the noise model.

Abstract translation: 一种用于均衡在加压空气输送系统内产生的语音信号的方法，所述方法包括以下步骤：基于吸入噪声产生吸入噪声模型（1152） 接收包括语音信号的输入信号（802）; 以及基于噪声模型均衡语音信号（1156）。

公开(公告)号：WO2006007342B1

公开(公告)日：2006-05-04

申请号：PCT/US2005020299

申请日：2005-06-09

Applicant: MOTOROLA INC ,  KUSHNER WILLIAM M ,  HARTON SARA M ,  JASIUK MARK A

Inventor： KUSHNER WILLIAM M ,  HARTON SARA M ,  JASIUK MARK A

IPC: G01L15/00 ,  G10L15/06 ,  G10L21/00

CPC classification number: G10L17/26 ,  A62B18/08 ,  B63C11/26 ,  G10L15/20 ,  G10L21/0208

Abstract: A method for characterizing inhalation noise within a pressurized air delivery system, the method including the steps of: generating an inhalation noise model (912, 1012) based on inhalation noise; receiving an input signal (802) that includes inhalation noise comprising at least one inhalation noise burst; comparing (810) the input signal to the noise model to obtain a similarity measure; comparing the similarity measure to at least one threshold (832, 834) to detect the at least one inhalation noise burst; and characterizing (1354, 1356) the at least one detected inhalation noise burst.

Abstract translation: 一种用于表征加压空气输送系统内的吸入噪声的方法，所述方法包括以下步骤：基于吸入噪声产生吸入噪声模型（912,1012）; 接收包括至少一个吸入噪声突发的吸入噪声的输入信号（802）; 将输入信号与噪声模型进行比较（810）以获得相似性度量; 将所述相似性度量与至少一个阈值（832,834）进行比较以检测所述至少一个吸入噪声突发; 以及表征（1354,1356）所述至少一个检测到的吸入噪声突发。

公开(公告)号：EP0556354A4

公开(公告)日：1995-07-26

申请号：EP92917101

申请日：1992-08-03

Applicant: MOTOROLA INC

Inventor： GERSON IRA A ,  WINTER ERIC H ,  JASIUK MARK A

IPC: G10L19/04 ,  G10L19/00 ,  G10L19/06 ,  G10L19/12 ,  G10L19/14 ,  H03M13/35 ,  G10L5/00 ,  G06F11/00

CPC classification number: G10L19/06 ,  G10L19/135 ,  G10L19/18 ,  H03M13/35

Abstract: A method for protecting information bits wherein input data bits, at least some of which are to be protected, are sorted based upon information determined from a subset of the input data bits. An error control coding technique is applied to at least some of the sorted bits. In the preferred embodiment, an input data stream of voice coder bits is separated into arrays of bits. A first array (302) comprises voice coder bits needing error protection, with the bits arranged in order of importance determined by voicing mode. The second array (303) comprises bits that will not be error protected. The bits from the first array are provided to the input of an encoder (304), then the encoded bits are combined (305) with the bits from the second array (303) to form a bit stream.

Abstract translation: 一种用于保护信息比特的方法，其中输入数据比特（其中至少一些将被保护）基于从输入数据比特的子集确定的信息被分类。 对至少一些分类的比特应用差错控制编码技术。 在优选实施例中，话音编码器比特的输入数据流被分成比特阵列。 第一阵列（302）包括需要错误保护的语音编码器位，其中按重要性排列的位按发音模式确定。 第二阵列（303）包含不会被错误保护的位。 将来自第一阵列的比特提供给编码器（304）的输入，然后将编码比特与来自第二阵列（303）的比特组合（305）以形成比特流。