Low-cost filter coefficient determination method in reverberation removal
    41.
    发明专利
    Low-cost filter coefficient determination method in reverberation removal 有权
    低成本过滤器系统拆除中的系数确定方法

    公开(公告)号:JP2008058900A

    公开(公告)日:2008-03-13

    申请号:JP2006238873

    申请日:2006-09-04

    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

    Speech recording system, sound recording device, speech analyzing device, speech recording method, and program
    42.
    发明专利
    Speech recording system, sound recording device, speech analyzing device, speech recording method, and program 有权
    语音录音系统,声音记录设备,语音分析设备,语音记录方法和程序

    公开(公告)号:JP2005338402A

    公开(公告)日:2005-12-08

    申请号:JP2004156571

    申请日:2004-05-26

    CPC classification number: G10L21/028

    Abstract: PROBLEM TO BE SOLVED: To provide a method of specifying speakers of individual voices from recorded voices of a plurality of speakers with simple device constitution, and a system using the same method. SOLUTION: The system is equipped with microphones 10 which are provided by the speakers, a speech processing section 20 which imparts unique characteristics to speech signals of two channels recorded by the microphones 10 through mutually different speech processes and mixes the signals by the channels, and an analysis section 40 which takes analyses corresponding to the unique characteristics imparted to the speech signals by the microphones 10 through the processes of the speech process section 20 to specify speakers by utterance sections of the speech signals. COPYRIGHT: (C)2006,JPO&NCIPI

    Abstract translation: 要解决的问题:提供一种用简单的装置构成从多个扬声器的记录的声音指定各个声音的扬声器的方法,以及使用相同方法的系统。 解决方案:该系统配备有由扬声器提供的麦克风10,语音处理部分20,其通过相互不同的语音处理向麦克风10记录的两个声道的语音信号赋予独特的特性,并将信号混合在一起 频道,以及分析部分40,通过语音处理部分20的处理,通过麦克风10对语音信号赋予的独特特征进行分析,以通过语音信号的话语部分指定扬声器。 版权所有(C)2006,JPO&NCIPI

    Information processing device, large vocabulary continuous speech recognition method, and program
    47.
    发明专利
    Information processing device, large vocabulary continuous speech recognition method, and program 有权
    信息处理设备,大型语音连续语音识别方法和程序

    公开(公告)号:JP2013148697A

    公开(公告)日:2013-08-01

    申请号:JP2012008732

    申请日:2012-01-19

    CPC classification number: G10L15/04 G10L15/10 G10L2015/025

    Abstract: PROBLEM TO BE SOLVED: To provide a technique for performing speech recognition using an acoustic invariant structure for a large vocabulary continuous speech.SOLUTION: An information processing device 100 includes: a speech recognition processing unit for receiving a speech as input, performing speech recognition, and outputting speech recognition scores together with a plurality of hypotheses which result from the recognition; a structure score calculation unit for calculating a structure score that is a score obtained for each hypothesis by considering all phoneme pairs constituting the hypothesis and summing up phoneme pair inter-distribution distance likelihoods multiplied by phoneme pair-by-pair weights; and a ranking unit for re-ranking the multiple hypotheses on the basis of the sum of the speech recognition score and the structure score.

    Abstract translation: 要解决的问题:提供一种用于使用用于大词汇连续语音的声学不变结构来执行语音识别的技术。解决方案:信息处理设备100包括:语音识别处理单元,用于接收语音作为输入,执行语音识别, 并输出语音识别分数以及由识别产生的多个假设; 结构分数计算单元,用于通过考虑构成假设的所有音素对并且将音素对分配间距可能性乘以音素对逐对加权来计算作为对于每个假设获得的得分的结构得分; 以及基于语音识别分数和结构得分的总和重新排列多个假设的排名单位。

    Method and system for position detection of sound source
    48.
    发明专利
    Method and system for position detection of sound source 有权
    用于位置检测声源的方法和系统

    公开(公告)号:JP2010021854A

    公开(公告)日:2010-01-28

    申请号:JP2008181514

    申请日:2008-07-11

    CPC classification number: G01S5/30 H04R3/005

    Abstract: PROBLEM TO BE SOLVED: To provide a method and system for detecting a position of a user of a home television game machine. SOLUTION: A speaker 506 mounted in a remote controller is used to reproduce a signal of a predetermined reproduced sound, the reproduced sound is observed respectively by two microphones properly provided in the vicinity of a television screen, CSP (while mutual correlation) coefficients of a signal of an observation sound respectively observed and the signal of the reproduced sound are calculated, and distances between the speaker inside the remote controller and the microphones are calculated, thereby acquiring longitudinal and lateral absolute positions of the remote controller with respect to a microphone array. An interference sound of an environmental sound or noise is canceled by the correlation calculation. COPYRIGHT: (C)2010,JPO&INPIT

    Abstract translation: 要解决的问题:提供一种用于检测家庭电视游戏机的用户的位置的方法和系统。 解决方案:安装在遥控器中的扬声器506用于再现预定再现声音的信号,再现的声音分别由在电视机屏幕附近适当提供的两个麦克风CSS(相互相关) 分别观察观测声音的信号的系数和再生声音的信号,并且计算遥控器内的扬声器与麦克风之间的距离,从而获得遥控器相对于扬声器的纵向和横向的绝对位置 麦克风阵列 通过相关计算来消除环境声音或噪声的干扰声。 版权所有(C)2010,JPO&INPIT

    Speech recognition and synthesis system, program and method
    49.
    发明专利
    Speech recognition and synthesis system, program and method 有权
    语音识别和合成系统,程序和方法

    公开(公告)号:JP2009282330A

    公开(公告)日:2009-12-03

    申请号:JP2008134759

    申请日:2008-05-22

    Abstract: PROBLEM TO BE SOLVED: To provide a method, a means and a program for high accuracy speech recognition and naturally synthesized speech, output in a language having large variations in the speech tone. SOLUTION: A statistic model is learned, by observing F0 tilt by using a linear approximation method or a global smoothing method, of F0 of a start point and an end point of a phoneme, and the F0 tilt is evaluated in runtime, and synthesis speech in which the F0 is corrected, based on cost calculation is output. Time change of the F0 tilt in a syllable is modeled, by learning a decision tree for each region into which the syllable is suitably and equally divided. Likelihood is evaluated by estimating an error range in the observed F0 tilt. By linking these operations, high-accuracy speech recognition and natural tone synthesis speech output are obtained. COPYRIGHT: (C)2010,JPO&INPIT

    Abstract translation: 要解决的问题:提供用于高精度语音识别和自然合成语音的方法,装置和程序,以具有较大变化的语言语言输出。

    解决方案:通过使用线性近似法或全局平滑方法,通过观察起始点和音素的终点的F0的F0倾斜来学习统计模型,并且在运行时评估F0倾斜度, 并输出基于成本计算校正F0的合成语音。 音节F0倾斜的时间变化是通过学习一个决策树,为每个区域进行适当和均等分割的音节。 通过估计观察到的F0倾斜的误差范围来评估似然。 通过链接这些操作,获得高精度语音识别和自然音合成语音输出。 版权所有(C)2010,JPO&INPIT

    Voice processing system, method and program
    50.
    发明专利
    Voice processing system, method and program 有权
    语音处理系统,方法和程序

    公开(公告)号:JP2009058708A

    公开(公告)日:2009-03-19

    申请号:JP2007225195

    申请日:2007-08-31

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

    Abstract: PROBLEM TO BE SOLVED: To provide a voice processing technique attaining stable voice recognition even in noise. SOLUTION: A high-order term and a low-order term of cepstrum of an observation voice are cut to design a filter directly from the observation voice itself. The filter is thereby made a filter with weight at a harmonic structure part in a section of a voiced sound, and a filter close to flat in a section of voiceless sound without the harmonic structure. Since this change is continuous, stable processing can be performed without distinguishing the voiced sound section from the voiceless sound section. COPYRIGHT: (C)2009,JPO&INPIT

    Abstract translation: 要解决的问题:提供甚至在噪声中实现稳定语音识别的语音处理技术。 解决方案:切割观察语音的倒谱的高阶项和低阶项,直接从观察声音本身设计滤波器。 由此,过滤器在声音的一部分中的谐波结构部分处具有重量的滤波器,并且在没有谐波结构的无声声音部分中接近平坦的滤波器。 由于该变化是连续的,因此可以进行稳定的处理,而不区分浊音部分与无声音部分。 版权所有(C)2009,JPO&INPIT

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