公开(公告)号：US20100063817A1

公开(公告)日：2010-03-11

申请号：US12531219

申请日：2007-03-14

申请人： Soichi Toyama ,  Ikuo Fujita ,  Yukio Kamoshida

发明人： Soichi Toyama ,  Ikuo Fujita ,  Yukio Kamoshida

IPC分类号： G10L15/06

CPC分类号： G10L17/04

摘要： An acoustic model registration apparatus, an talker recognition apparatus, an acoustic model registration method and an acoustic model registration processing program, each of which prevents certainly an acoustic model having a low recognition capability for talker from being registered certainly, are provided.When a talker utters for the N utterances and the utterance sounds of the N utterances are input through the microphone 1, the sound feature quantity extraction part 4 extracts sound feature quantities which indicate the acoustic features of the input utterance sounds, wherein each sound feature quantity has one-to-one correspondence to each utterance, the talker model generation part 5 generates a talker model based on the extracted sound feature quantities for the N utterances, the collation part 6 calculates the degree of individual similarity between the each sound feature quantity of the N utterances and the talker model generated above, and only in the case that all the calculated degrees of similarities of the N utterances are equal to or more than the threshold value, the similarity verifying part 9 directs to register the generated talker model in the talker models' database as a talker model for the talker recognition.

摘要翻译： 提供了一种声学模型登记装置，讲话者识别装置，声学模型登记方法和声学模型登记处理程序，其中，每个都可以防止当前登记具有低的识别能力的声学模型被登记。 当讲话者发出N个话语时，通过麦克风1输入N个发音的发声，声音特征量提取部分4提取指示输入话音的声音特征的声音特征量，其中每个声音特征量 讲话者模型生成部5基于所提取的N个特征量的声音特征量生成说话者模型，计算出每个声音特征量之间的个体相似度的程度， 上面产生的N个话语和说话者模型，并且只有在所有计算出的N个话语的相似程度等于或大于阈值的情况下，相似性验证部分9才指示将所产生的讲话者模型注册在 讲话者模型的数据库作为谈话者识别的谈话者模型。

公开(公告)号：US07177809B2

公开(公告)日：2007-02-13

申请号：US10164600

申请日：2002-06-10

申请人： Takehiko Shioda ,  Soichi Toyama ,  Keiichi Yamauchi ,  Hiroaki Shibasaki ,  Hideki Amano

发明人： Takehiko Shioda ,  Soichi Toyama ,  Keiichi Yamauchi ,  Hiroaki Shibasaki ,  Hideki Amano

IPC分类号： G10L15/00

CPC分类号： G10L17/26 ,  G10L15/26

摘要： A contents presenting system includes: an analyzing unit which collects and analyzes user's conversation to output an analysis result; a contents acquiring unit which acquires contents from a contents database based on the analysis result; and a contents presenting unit which presents the acquired contents to the user. Since the analysis result of user's conversation includes a factor representing the environment where the user is talking, by determining contents based on the analysis result, it is possible to provide the contents which is suited for the environment where the user is present.

摘要翻译： 内容呈现系统包括：分析单元，其收集和分析用户的对话以输出分析结果; 内容获取单元，其基于分析结果从内容数据库获取内容; 以及将所获取的内容呈现给用户的内容呈现单元。 由于用户对话的分析结果包括表示用户正在通话的环境的因素，通过基于分析结果确定内容，可以提供适合于存在用户的环境的内容。

公开(公告)号：US06629283B1

公开(公告)日：2003-09-30

申请号：US09671278

申请日：2000-09-27

申请人： Soichi Toyama

发明人： Soichi Toyama

IPC分类号： G06F1100

CPC分类号： G10L19/005

摘要： A quantization error correcting device corrects quantization error included in audio information at the time of decoding. The audio information is divided into a plurality of frequency bands and compressive-encoded for each frequency band with bit allocation determined based on audible frequency characteristic. The device includes: a detecting unit for detecting, based on bit allocation information indicating bit allocation and encoded values of the compressive-encoded audio information, a range of quantization error indicating a range in which audio information value before compressive-encoding corresponding to the encoded value exists; and an outputting unit for outputting a decoded value corresponding to one of the encoded values based on the detected range of quantization error and the ranges of quantization errors of other correlated ones of the encoded values.

摘要翻译： 量化误差校正装置在解码时校正音频信息中包含的量化误差。 音频信息被分成多个频带，并且对于每个频带进行压缩编码，其中位分配基于可听频率特性确定。 该装置包括：检测单元，用于基于指示比特分配的比特分配信息和压缩编码音频信息的编码值来检测量化误差的范围，该范围指示与编码的对应的压缩编码之前的音频信息值的范围 价值存在; 以及输出单元，用于基于所检测的量化误差的范围和编码值中的其他相关编码值的量化误差的范围来输出与编码值之一相对应的解码值。

公开(公告)号：US5578948A

公开(公告)日：1996-11-26

申请号：US819987

申请日：1992-01-13

申请人： Soichi Toyama

发明人： Soichi Toyama

IPC分类号： G10H1/12 ,  G10H1/16 ,  G10K15/04 ,  H03B19/00 ,  H03B28/00 ,  G10H7/00

CPC分类号： H03B19/00 ,  G10H1/125 ,  G10H1/16 ,  G10H2250/545 ,  H03B28/00

摘要： A harmonic tone generator produces a harmonics signal even for input audio signals of a small amplitude. Conversion of a digitized audio signal in accordance with a predetermined non-linear function is performed also for an audio signal of a small amplitude. According to the second aspect of the invention, a level difference between the digital audio signal level in the present sampling time and the audio signal level in the preceding sampling time is detected and the detected level difference is converted to an output value in accordance with a predetermined non-linear function by a non-linear converting circuit. The converted output value is accumulated. According to the third aspect of the invention, the detected level difference is converted to a function conversion output in accordance with a predetermined function by a non-linear converting circuit. A gain of an amplifier to amplify the audio signal in the present sampling time is changed in accordance with the function conversion output.

摘要翻译： 即使对于小幅度的输入音频信号，谐波发生器也产生谐波信号。 对于小幅度的音频信号也执行根据预定非线性函数的数字化音频信号的转换。 根据本发明的第二方面，检测当前采样时间的数字音频信号电平与先前采样时间中的音频信号电平之间的电平差，并将检测到的电平差转换为根据 通过非线性转换电路预定的非线性函数。 转换的输出值被累加。 根据本发明的第三方面，通过非线性转换电路根据预定的功能将检测到的电平差转换为功能转换输出。 根据功能转换输出，改变放大器在当前采样时间放大音频信号的增益。

公开(公告)号：US5442711A

公开(公告)日：1995-08-15

申请号：US275243

申请日：1994-07-15

申请人： Soichi Toyama

发明人： Soichi Toyama

IPC分类号： G10K15/12 ,  H03G3/00 ,  H04B1/00

CPC分类号： G10K15/12 ,  Y10S84/26

摘要： A sound echo machine as an acoustic signal processing unit of the present invention comprising an adder to which an input signal is fed, and a delay circuit for delaying the signal fed from the adder for a certain time to repeatedly feed back to the adder to generate an echo sound further comprises an input signal level detector for detecting the level of the input signal and sending it to a frequency oscillator to vary the oscillating frequency in accordance with the thus detected signal level for feeding it later to the delay circuit so as to modulate the time to be delayed at a predetermined cycle, whereby it can create an acoustic field in which a listener can feel as if various level of reflected sounds are coming towards him from various directions. On the other hand, a sound effecter as an acoustic signal processing unit comprising a plurality of acoustic signal processing sections, a plurality of attenuators each connected to these acoustic signal processing sections, and an adder for summing up all the signals from these attenuators further comprises a signal mixing ratio control section for monitoring the input acoustic signal level, and determining a signal mixing ratio among the respective output signals from the plurality of acoustic signal processing sections in accordance with the thus monitored level of the input acoustic signal, whereby even a simple structure can provide a specific sound effect.

公开(公告)号：US07979718B2

公开(公告)日：2011-07-12

申请号：US11910415

申请日：2006-03-24

申请人： Soichi Toyama ,  Ikuo Fujita ,  Mitsuya Komamura

发明人： Soichi Toyama ,  Ikuo Fujita ,  Mitsuya Komamura

IPC分类号： H04K1/00 ,  H04L9/00

CPC分类号： G10L17/16 ,  G10L17/10

摘要： An operator recognition device is provided that eliminates the registration of data such as HMM data having a characteristic amount for which error in recognition occurs easily when recognizing an operator, and thus reduces the possibility of errors in recognition, and has stable recognition performance. When registering HMM data that is used when performing recognition processing, a speaker recognition device 100 eliminates the registration of HMM data of a password having a characteristic amount of the spoken voice component that is similar to a characteristic amount that is indicated by HMM data that is already registered, and does not allow the registration of HMM data for which it is estimated that error in recognition will occur easily during the recognition process.

摘要翻译： 提供了一种操作者识别装置，其消除了在识别操作者时容易识别出具有识别错误的特征量的HMM数据的登记，从而降低识别错误的可能性，并且具有稳定的识别性能。 当注册执行识别处理时使用的HMM数据时，说话人识别装置100消除了具有与由HMM数据表示的特征量相似的口语语音成分的特征量的密码的HMM数据的注册， 已经注册，并且不允许HMM数据的注册，估计在识别过程中容易发生识别错误。

公开(公告)号：US20090106025A1

公开(公告)日：2009-04-23

申请号：US12293943

申请日：2007-03-16

申请人： Soichi Toyama

发明人： Soichi Toyama

IPC分类号： G10L17/00

CPC分类号： G10L17/04

摘要： EN) A speaker recognition system (1) includes a speaker model registration device (10) which registers a speaker model for speaker recognition in the speaker recognition system. The speaker model registration device includes acquisition means (13) for acquiring utterances by n+α times (wherein n is an integer not smaller than 2 and α is an integer not smaller than 1); calculation means (20) for calculating a speaker model by using the acquired utterances of n times as utterances for registration; correlation means (30) for correlating the calculated speaker model by using the acquired utterances of α times as correlation utterances; and registration means (40) for registering those having the correlation result satisfying a predetermined reference among the correlated speaker models, as the speaker model for speaker recognition.

摘要翻译： 说明书识别系统（1）包括扬声器模型登记装置（10），其在扬声器识别系统中登记用于说话者识别的扬声器模型。 扬声器模型登记装置包括用于通过n +α次获取话语的获取装置（13）（其中n是不小于2的整数，α是不小于1的整数）; 计算装置（20），用于通过使用所获取的n次作为用于注册的话语的话语来计算说话者模型; 相关装置（30），用于通过使用获得的α时间的话语作为相关语句来将计算出的说话者模型相关; 以及用于将具有相关结果满足预定参考的那些在相关的说话者模型中登记的登记装置（40）作为用于说话者识别的说话者模型。

公开(公告)号：US20080046844A1

公开(公告)日：2008-02-21

申请号：US11547053

申请日：2005-03-31

申请人： Shinichi Sugie ,  Naohiko Ichihara ,  Yuji Koga ,  Soichi Toyama

发明人： Shinichi Sugie ,  Naohiko Ichihara ,  Yuji Koga ,  Soichi Toyama

IPC分类号： G06F3/048

CPC分类号： G01C21/3611 ,  G06F16/29 ,  G08G1/096716 ,  G08G1/09675 ,  G08G1/096791

摘要： The present invention is directed to provide a data selecting apparatus and a navigation apparatus capable of easily and promptly selecting one piece of data from a plurality of pieces of data. A navigation apparatus 100 has: a display controller 111 for obtaining name data and genre information of each point data from a map data storing unit 105, generating display data for displaying names of the point data, which are arranged by the genre information at the same hierarchical level on the basis of the obtained name data and genre information, and performing the display control on the generated display data, and an operating unit 106 used for selecting a genre to which point data to be selected by the user belongs and selecting a name of one piece of the point data from the selected genre. The display controller 111 performs generation of display data for displaying the genre selected by the operating unit 106 and display control on the display data or performs generation of display data at the time when the name of point data is selected by the operating unit 106 and display control on the display data interlockingly with selecting operation executed by using the operating unit 106.

摘要翻译： 本发明旨在提供一种数据选择装置和导航装置，其能够容易且迅速地从多条数据中选择一条数据。 导航装置100具有：显示控制器111，用于从地图数据存储单元105获取每个点数据的名称数据和类型信息，生成用于显示由相同类型的信息排列的点数据的名称的显示数据 基于所获得的名称数据和类型信息进行分级，并对生成的显示数据执行显示控制;以及操作单元106，用于选择要由用户选择的点数据所属的类型，并选择名称 来自所选类型的一个点数据。 显示控制器111执行用于显示由操作单元106选择的类型的显示数据的生成和对显示数据的显示控制，或者在操作单元106选择点数据的名称时执行显示数据的生成，并且显示 与通过使用操作单元106执行的选择操作互动地控制显示数据。

公开(公告)号：US20070203700A1

公开(公告)日：2007-08-30

申请号：US11547083

申请日：2005-03-22

申请人： Soichi Toyama

发明人： Soichi Toyama

IPC分类号： G10L15/04

CPC分类号： G10L15/08 ,  G10L15/142 ,  G10L15/187

摘要： A speech recognition apparatus and speech recognition method are provided for reducing such events as erroneous recognition and disabled recognition and improving a recognition efficiency. The speech recognition apparatus generates a word model based on a dictionary memory and a sub-word sound model, and matches the word model with a speech input signal in accordance with a predetermined algorithm to perform a speech recognition for the speech input signal, wherein the apparatus comprises main matching means, operative when matching the word model with the speech input signal along a processing path indicated by the algorithm, for limiting the processing path based on a course command to select the word model most approximate to the speech input signal, local template storing means for previously typifying local sound features of spoken speeches for storage as local templates; and local matching means for matching each of component sections of the speech input signal with the local templates stored in the local template storing means to definitely determine a sound feature for each of the component sections, and generating the course command in accordance with the result of the definite determination.

摘要翻译： 提供语音识别装置和语音识别方法，用于减少错误识别和无效识别等事件，提高识别效率。 语音识别装置根据字典存储器和子字声音模型生成字模型，并根据预定算法将字模型与语音输入信号进行匹配，以对语音输入信号执行语音识别，其中， 装置包括主要匹配装置，当沿着算法指示的处理路径与单词模型与语音输入信号相匹配时，主要匹配装置用于基于课程命令限制处理路径，以选择最接近于语音输入信号的单词模型，局部 模板存储装置，用于预先表示作为本地模板存储的口头演讲的本地声音特征; 以及本地匹配装置，用于将语音输入信号的每个分量部分与存储在本地模板存储装置中的本地模板进行匹配，以确定每个分量部分的声音特征，并根据结果 明确的决定。

公开(公告)号：US07130799B1

公开(公告)日：2006-10-31

申请号：US09684331

申请日：2000-10-10

申请人： Katsumi Amano ,  Shisei Cho ,  Soichi Toyama ,  Hiroyuki Ishihara

发明人： Katsumi Amano ,  Shisei Cho ,  Soichi Toyama ,  Hiroyuki Ishihara

IPC分类号： G10L13/00

CPC分类号： G10L13/04 ,  G10L13/07

摘要： A speech synthesizing method which synthesizes speech naturally is disclosed. Standardized frame power values of an n-th frame is calculated when frame power values at head and tail frames in a phoneme are standardized. An average value of the power values sampled from the power frequency characteristics in the n-th frame at a predetermined frequency interval is set as a mean frame power value. A sum of squares of signal levels in one frame of a frequency signal from a sound source is calculated as a frame power correction value. A speech envelope signal is calculated as a function having variables of the standardized frame power values, the frame power correction value and the mean frame power value. The speech envelope signal adjusts the amplitude level of a speech waveform signal supplied from a vocal tract filter according to the level of the speech envelope signal.

摘要翻译： 公开了一种自然合成语音的语音合成方法。 在音素中的头部和尾部帧的帧功率值被标准化时，计算第n帧的标准化帧功率值。 以预定频率间隔从第n帧中的功率频率特性采样的功率值的平均值被设置为平均帧功率值。 将来自声源的频率信号的一帧中的信号电平的平方和计算为帧功率校正值。 计算语音包络信号作为具有标准化帧功率值，帧功率校正值和平均帧功率值的变量的函数。 语音包络信号根据语音包络信号的电平来调节从声道滤波器提供的语音波形信号的幅度电平。