公开(公告)号：JP2011113044A

公开(公告)日：2011-06-09

申请号：JP2009271890

申请日：2009-11-30

Applicant: Internatl Business Mach Corp ,  インターナショナル・ビジネス・マシーンズ・コーポレーションInternational Business Maschines Corporation

Inventor： ICHIKAWA OSAMU ,  FUKUDA TAKASHI ,  NISHIMURA MASAFUMI

IPC: G10L15/28 ,  G10L15/20 ,  G10L21/02

CPC classification number: G10L25/78 ,  G10L15/20 ,  G10L21/028 ,  G10L2021/02166

Abstract: PROBLEM TO BE SOLVED: To provide technology for extracting objective voice by efficiently suppressing mixing of other voice than objective voice, in a plurality pieces of voice which come from different directions.
SOLUTION: The objective voice is extracted by performing at least either gain adjustment processing and segmentation processing of an utterance section, on a voice signal obtained by each of first and second voice input units which are arranged with a predetermined distance apart, by using a weighted Cross-Power Spectrum Phase (CSP) coefficient which becomes a small value in a frequency band which is likely to be influenced by other voice than the objective voice.
COPYRIGHT: (C)2011,JPO&INPIT

Abstract translation: 要解决的问题：通过有效地抑制来自不同方向的多个声音中的其他声音的混合而不是客观语音来提取目标声音的技术。 解决方案：通过对由发声部分进行的增益调整处理和分段处理，对由第一和第二语音输入单元中的每一个以预定距离间隔排列而获得的语音信号进行至少一个提取，目标声音由 使用加权的跨功率谱相位（CSP）系数，该系数在可能受到客观声音的其他声音影响的频带中变成小值。 版权所有（C）2011，JPO＆INPIT

公开(公告)号：JP2013178575A

公开(公告)日：2013-09-09

申请号：JP2013109608

申请日：2013-05-24

Applicant: Internatl Business Mach Corp ,  インターナショナル・ビジネス・マシーンズ・コーポレーションInternational Business Machines Corporation

Inventor： ICHIKAWA OSAMU ,  FUKUDA TAKASHI ,  NISHIMURA MASAFUMI

IPC: G10L15/02 ,  G10L25/24

CPC classification number: G10L15/02 ,  G10L15/20 ,  G10L25/24

Abstract: PROBLEM TO BE SOLVED: To provide a technique for extracting features even more robust to reverberations, noises, and the like from a speech signal.SOLUTION: A speech feature extraction apparatus is configured to: receive, as an input, values obtained by adding a spectrum of each frame of a speech signal segmented into frames to an average spectrum that is the average of spectra over all frames that are overall speech; and, for each frame, multiply said values by weights of a mel filter bank to sum up the products, apply the discrete cosine transform to the logarithm of the sum, and calculate, and define as a delta feature, the difference in the discrete cosine transform between former and later frames.

Abstract translation: 要解决的问题：提供一种用于从语音信号中提取对于混响，噪声等更加鲁棒的特征的技术。解决方案：语音特征提取装置被配置为：作为输入接收通过添加 将分割成帧的语音信号的每个帧的频谱分解为平均频谱，该平均频谱是作为整个语音的所有帧的频谱的平均值; 并且，对于每个帧，将所述值乘以呃滤波器组的权重以对产物进行求和，将离散余弦变换应用于和的对数，并计算并定义为离散余弦差的Δ特征 在前后帧之间进行转换。

公开(公告)号：JP2012032557A

公开(公告)日：2012-02-16

申请号：JP2010171278

申请日：2010-07-30

Applicant: Internatl Business Mach Corp ,  インターナショナル・ビジネス・マシーンズ・コーポレーションInternational Business Maschines Corporation

Inventor： FUKUDA TAKASHI ,  NISHIMURA MASAFUMI

IPC: G10L15/10 ,  G10L15/04

Abstract: PROBLEM TO BE SOLVED: To provide a technology capable of detecting an ingressive in a voice signal with a high detection rate and a high degree of accuracy.SOLUTION: An ingressive detection device refers to each acoustic model of ingressive and non-ingressive for determining an ingressive candidate and generates a feature vector with setting simplex information meaning information on ingressive candidate simplex, and context information as an element. The context information means information on a relation between the ingressive candidate and a speech section including the ingressive candidate, a relation between the ingressive candidate and an ingressive candidate before and after the ingressive candidate or both relations. The ingressive detection device obtains classification reference information for classifying the ingressive candidate into either the ingressive or the non-ingressive, through machine learning with setting the feature vector as input, and classifies the ingressive candidate into either the ingressive or the non-ingressive based on the classification reference information.

Abstract translation: 要解决的问题：提供能够以高检测率和高精度检测语音信号中的入侵的技术。 入侵检测装置是指入侵性和非侵入性的每个声学模型，用于确定入侵候选，并且生成特征向量，其中设置单数信息意味着入侵候选单形的信息和上下文信息作为元素。 上下文信息是指关于入侵候选者和包括入境候选人的语音部分之间的关​​系的信息，入境候选人之间和入侵候选人之间的关系以及两者之间的关系。 入侵检测装置通过设置特征向量作为输入，通过机器学习获得入侵候选分类为入侵或非入侵的分类参考信息，并将入侵候选分类为入侵或非进入基于 分类参考信息。 版权所有（C）2012，JPO＆INPIT

公开(公告)号：JP2008275881A

公开(公告)日：2008-11-13

申请号：JP2007119194

申请日：2007-04-27

Applicant: Internatl Business Mach Corp ,  インターナショナル・ビジネス・マシーンズ・コーポレーションInternational Business Maschines Corporation

Inventor： FUKUDA TAKASHI ,  ICHIKAWA OSAMU ,  NISHIMURA MASAFUMI

IPC: G10L15/20 ,  G10L15/28 ,  H04R3/00

CPC classification number: G10L15/20 ,  G10L15/02 ,  G10L21/02 ,  G10L21/0272 ,  G10L2021/02161

Abstract: PROBLEM TO BE SOLVED: To extract only voice of a target person under noise environment, without requiring a large scale microphone array and a reference signal of noise.
SOLUTION: An object sound extraction method is disclosed in which a practical speech recognition performance is actualized only by performing gain adjustment between spectrum subtraction (SS) processing and flooring processing, as processing for two channel input speech which is obtained from the microphones 1 and 2 etc. As the gain adjustment, a CSP (Cross-power Spectrum Phase) coefficient, which is cross-correlation between two channel signals, can be utilized. In an indoor environment including a vehicle where audio background sound etc., a recognition rate of a voice command in a car navigation system is improved, then, usability of a speaker such as a driver is improved.
COPYRIGHT: (C)2009,JPO&INPIT

Abstract translation: 要解决的问题：在噪声环境下仅提取目标人员的声音，而不需要大规模的麦克风阵列和噪声的参考信号。 解决方案：公开了一种对象声音提取方法，其中仅通过在频谱减法（SS）处理和地板处理之间进行增益调整来实现实际语音识别性能，作为从麦克风获得的两声道输入语音的处理 1和2等。作为增益调整，可以使用在两个信道信号之间互相关的CSP（跨功率谱相位）系数。 在包括音响背景音等的车辆的室内环境中，提高了汽车导航系统中的语音命令的识别率，因此提高了诸如驾驶员的扬声器的可用性。 版权所有（C）2009，JPO＆INPIT

公开(公告)号：JP2008058900A

公开(公告)日：2008-03-13

申请号：JP2006238873

申请日：2006-09-04

Applicant: Internatl Business Mach Corp ,  インターナショナル・ビジネス・マシーンズ・コーポレーションInternational Business Maschines Corporation

Inventor： ICHIKAWA OSAMU ,  FUKUDA TAKASHI ,  NISHIMURA MASAFUMI

IPC: G10L15/20 ,  G10L21/02

CPC classification number: G10L2021/02082

Abstract: PROBLEM TO BE SOLVED: To solve the problem wherein although the performance of a voice recognition device deteriorates significantly in the circumstances in which there exists long reverberation, which is generally known, and most of the conventional reverberation removal methods require a large amount of calculation is not large, or for those where the amount of calculation is not large, some kind of previous knowledge (reverberation time of a room, etc.) is required.
SOLUTION: The coefficient determination in the conventional techniques, in which the multiple value of the coefficient of power spectrum of the past frame is subtracted from the power spectrum of the current frame is calculated at low cost, without having to use the information that incurs calculation cost, such as acoustic model or multi-channel input. As a specific method, a voice power track that properly follows the frame of large power and follows the frame of small power late is obtained, and the interval of which the voice power differs significantly from the voice power of the current frame that is smoothed in the time direction is deduced as being an utterance terminal reverberation interval, and the filter coefficient is decided, in such a manner as to minimize the weighted total sum of the residual voice power in the interval and the subtracted power in the utterance interval (not including the reverberation interval).
COPYRIGHT: (C)2008,JPO&INPIT

Abstract translation: 要解决的问题为了解决这样一个问题，即在通常已知的存在长混响的情况下语音识别装置的性能显着恶化，并且大多数传统的混响消除方法需要大量的 计算量不大，或对于计算量不大的情况，需要某种以前的知识（房间的混响时间等）。 解决方案：以低成本计算过去帧的功率谱系数的多个值从当前帧的功率谱中减去的常规技术中的系数确定，而不必使用该信息 导致计算成本，如声学模型或多通道输入。 作为具体的方法，获得了正确跟随大功率帧并且跟随小功率帧的语音功率轨迹，并且语音功率的间隔与当前平滑化的帧的语音功率显着不同 将时间方向推定为发声终端混响间隔，并且以使得间隔中的剩余语音功率的加权总和和话音间隔中的减法功率（不包括）的方式来决定滤波器系数 混响间隔）。 版权所有（C）2008，JPO＆INPIT

公开(公告)号：JP2010026361A

公开(公告)日：2010-02-04

申请号：JP2008189504

申请日：2008-07-23

Applicant: Internatl Business Mach Corp ,  インターナショナル・ビジネス・マシーンズ・コーポレーションInternational Business Maschines Corporation

Inventor： FUKUDA TAKASHI ,  ICHIKAWA OSAMU ,  NISHIMURA MASAFUMI

IPC: G10L21/02 ,  G10L15/00 ,  H04R1/40 ,  H04R3/00

Abstract: PROBLEM TO BE SOLVED: To accurately collect speech of only a specified speaker such as a sales person in counter selling or the like. SOLUTION: A speech collection system 10 extracts and collects target speech which is a target in a plurality of pieces of speech in which coming directions are different from each other. The system includes a microphone array 11 including at least first and second microphones 11a and 11b, in which the first and second microphones are arranged by separating them with a predetermined distance. Discrete Fourier transform is performed on each signal of speech received by the first and second microphones, and a plurality of cross spectrum power (CSP) coefficients related to the coming direction of speech are calculated, and a plurality of speech signals are detected from the plurality of CSP coefficients. Then, a speech direction index defined according to an angle between a line for connecting the first and second microphones and the coming direction, is detected from the plurality of calculated CSP coefficients, and the signal of the target speech is extracted from the plurality of speech signals, which are detected from the detected speech direction index. COPYRIGHT: (C)2010,JPO&INPIT

Abstract translation: 要解决的问题：准确地收集只有指定的演讲者，如销售人员等的销售人员的演讲。 解决方案：语音收集系统10提取和收集作为来自不同方向不同的多个语音中的目标的目标语音。 该系统包括麦克风阵列11，其包括至少第一麦克风11a和第二麦克风11b，其中第一和第二麦克风通过以预定距离分离而布置。 对由第一和第二麦克风接收的每个语音信号执行离散傅立叶变换，并且计算与语音的未来方向相关的多个交叉频谱功率（CSP）系数，并且从多个检测到多个语音信号 的CSP系数。 然后，从多个计算出的CSP系数中检测出根据用于连接第一和第二麦克风的线路之间的角度和来往方向所定义的语音方向索引，并且从多个语音中提取目标语音的信号 信号，其从检测到的语音方向索引检测。 版权所有（C）2010，JPO＆INPIT

公开(公告)号：JP2008166914A

公开(公告)日：2008-07-17

申请号：JP2006351358

申请日：2006-12-27

Applicant: Internatl Business Mach Corp ,  インターナショナル・ビジネス・マシーンズ・コーポレーションInternational Business Maschines Corporation

Inventor： SATO DAISUKE ,  FUKUDA TAKASHI

IPC: G06F17/30 ,  H04N7/173 ,  H04N21/433 ,  H04N21/435 ,  H04N21/438

CPC classification number: G06F17/30796 ,  G06F17/30265 ,  G11B27/10 ,  Y10S707/913

Abstract: PROBLEM TO BE SOLVED: To provide an apparatus for synchronizing a content data stream with meta data by the featured value of the content data stream.
SOLUTION: The apparatus for synchronizing content data with the meta data includes: a storage device wherein the meta data including the featured value of the content data are recorded; a means for calculating the featured value from the content data; a means for retrieving the meta data in the storage device on the basis of the calculated featured value; and a means for reproducing the retrieved meta data in synchronism with the content data.
COPYRIGHT: (C)2008,JPO&INPIT

Abstract translation: 要解决的问题：提供一种通过内容数据流的特征值使内容数据流与元数据同步的装置。 解决方案：用于将内容数据与元数据同步的装置包括：存储装置，其中记录包括内容数据的特征值的元数据; 用于从所述内容数据计算所述特征值的装置; 基于所计算的特征值来检索存储装置中的元数据的装置; 以及用于与内容数据同步地再现所检索的元数据的装置。 版权所有（C）2008，JPO＆INPIT

公开(公告)号：JP2009210617A

公开(公告)日：2009-09-17

申请号：JP2008050537

申请日：2008-02-29

Applicant: Internatl Business Mach Corp ,  インターナショナル・ビジネス・マシーンズ・コーポレーションInternational Business Maschines Corporation

Inventor： FUKUDA TAKASHI ,  ICHIKAWA OSAMU ,  NISHIMURA MASAFUMI

IPC: G10L11/02 ,  G10L11/00 ,  G10L15/04

CPC classification number: G10L25/93

Abstract: PROBLEM TO BE SOLVED: To provide a highly accurate voice activity detection method in a low S/N environment.
SOLUTION: The voice activity is performed by extracting a long-term spectrum variation component and a harmonic structure as feature vectors from a speech signal and increasing difference in feature vectors between speech and non-speech included in the speech signal by using the long-term spectrum variation component feature, or a long-term spectrum variation component extraction and a harmonic structure feature extraction. A correct rate and an accuracy rate of the voice activity detection is improved over conventional methods by using a long-term spectrum variation component having a window length over an average phoneme duration of an utterance in the speech signal. The voice activity detection system and method provides speech processing, automatic speech recognition, and speech output capable of very accurate voice activity detection.
COPYRIGHT: (C)2009,JPO&INPIT

Abstract translation: 要解决的问题：在低S / N环境中提供高精度的语音活动检测方法。 解决方案：通过从语音信号中提取长期频谱变化分量和谐波结构作为特征向量并且通过使用语音信号增加语音信号中包括的语音和非语音之间的特征向量的差异来执行语音活动 长期光谱变化分量特征，或长期光谱变化分量提取和谐波结构特征提取。 通过使用具有在语音信号中的话语的平均音素持续时间上的窗口长度的长期频谱变化分量，语音活动检测的正确率和准确率比常规方法得到改进。 语音活动检测系统和方法提供能够进行非常精确的语音活动检测的语音处理，自动语音识别和语音输出。 版权所有（C）2009，JPO＆INPIT

公开(公告)号：JP2009122904A

公开(公告)日：2009-06-04

申请号：JP2007295377

申请日：2007-11-14

Applicant: Internatl Business Mach Corp ,  インターナショナル・ビジネス・マシーンズ・コーポレーションInternational Business Maschines Corporation

Inventor： FUKUDA TAKASHI ,  ICHIKAWA OSAMU

IPC: G06F17/30

Abstract: PROBLEM TO BE SOLVED: To increase convenience of a user visiting an island in a virtual space.
SOLUTION: In the virtual space comprising a plurality of islands, positions of the islands are two-dimensionally mapped preferably by multidimensional scaling such as Kruskal's method such that the relation of distance between a characteristic vector including information on a profile, taste or the like of the user and a characteristic vector including information on a profile, an event or the like of each island is maintained. Thus, a map server provides the user with the islands arranged in accordance with the characteristic vector of the user based on mapped information. Thereby, it is convenient for the user to visit the island suited to his or her taste, so that a use frequency of the virtual space is improved.
COPYRIGHT: (C)2009,JPO&INPIT

Abstract translation: 要解决的问题：为了增加在虚拟空间中访问岛屿的用户的便利性。 解决方案：在包括多个岛的虚拟空间中，岛的位置优选地通过诸如Kruskal方法的多维缩放来二维映射，使得包括关于轮廓，味道或风味的信息的特征向量之间的距离的关系 维护用户的类似以及包含关于每个岛屿的简档，事件等的信息的特征向量。 因此，地图服务器基于映射的信息向用户提供根据用户的特征向量排列的岛。 由此，便于用户访问适合自己的口味的岛屿，从而提高虚拟空间的使用频率。 版权所有（C）2009，JPO＆INPIT