公开(公告)号：US06405169B1

公开(公告)日：2002-06-11

申请号：US09325544

申请日：1999-06-04

申请人： Reishi Kondo ,  Yukio Mitome

发明人： Reishi Kondo ,  Yukio Mitome

IPC分类号： G10L1300

CPC分类号： G10L13/08 ,  G10L13/04

摘要： The invention provides a speech synthesis apparatus which can produce synthetic speech of a high quality with reduced distortion. To this end, upon production of synthetic speech based on prosodic information and phonological unit information, the prosodic information is modified using the phonological unit information, and duration length information and pitch pattern information of phonological units of the prosodic information and the phonological unit information are modified with each other. The speech synthesis apparatus includes a prosodic pattern production section for receiving utterance contents as an input thereto and producing a prosodic pattern, a phonological unit selection section for selecting phonological units based on the prosodic pattern, a prosody modification control section for searching the phonological unit information selected by the phonological unit selection section for a location for which modification to the prosodic pattern is required and outputting information of the location for the modification and contents of the modification, a prosody modification section for modifying the prosodic pattern based on the information of the location for the modification and the contents of the modification outputted from the prosody modification control section, and a waveform production section for producing synthetic speech based on the phonological unit information and the prosodic information modified by the prosody modification section using a phonological unit database.

摘要翻译： 本发明提供一种语音合成装置，其能够以较低的失真产生高质量的合成语音。 为此，在基于韵律信息和语音单元信息生成合成语音时，使用语音单元信息修改韵律信息，并且韵律信息和语音单元信息的语音单元的持续时间长度信息和音调模式信息是 互相修改。 语音合成装置包括：韵律模式生成部，其接收作为其输入的话语内容并产生韵律模式;基于韵律模式选择语音单元的语音单元选择部;用于搜索语音单元信息的韵律修改控制部 由音素单元选择部选择需要对韵律模式进行修改的位置，并且输出修改位置的信息和修改内容的信息;韵律修改部分，用于基于位置的信息修改韵律模式 用于从韵律修改控制部分输出的修改和修改内容，以及基于音韵单元信息和由韵律修改部分修改的韵律信息使用语音学的韵律信息来产生合成语音的波形产生部分 单位数据库。

公开(公告)号：US5677984A

公开(公告)日：1997-10-14

申请号：US392482

申请日：1995-02-23

申请人： Yukio Mitome

发明人： Yukio Mitome

IPC分类号： G01R23/16 ,  G10L11/00 ,  G10L7/06

CPC分类号： G10L25/48 ,  G01R23/16 ,  G10L25/24

摘要： Respective samples of input speech data are transformed by a discrete Fourier transformer to obtain a spectrum of the speech data. Simultaneously, values of times of the respective samples are multiplied with the input speech data by a multiplier and a differential spectrum is obtained by transforming a result of the multiplication by a discrete Fourier transformer. A real part of a value obtained by dividing the differential spectrum by the spectrum by a quotient real part calculator and the real part is inverse-transformed by a discrete inverse Fourier transformer. The result of the inverse-transformation is divided by the values of the times of the respective samples to obtain a time function corresponding to phase. On the other hand, a time function corresponding to a logarithmic amplitude spectrum is obtained from an output of the inverse-Fourier transformer by means of a logarithmic amplitude spectrum calculator. A complex cepstrum is obtained by adding the both time functions at respective times by an adder.

摘要翻译： 输入语音数据的各样本由离散傅立叶变换器变换以获得语音数据的频谱。 同时，通过乘法器将各个样本的时间值与输入的语音数据相乘，并且通过将离散傅里叶变换器的乘法结果变换来获得差分频谱。 通过商实数部分计算器和实部将差分光谱除以光谱获得的值的实部由逆离散付里叶变换器进行逆变换。 将逆变换的结果除以各个样本的时间值，以获得对应于相位的时间函数。 另一方面，通过对数幅度频谱计算器从傅立叶逆变换器的输出获得与对数振幅谱相对应的时间函数。 通过加法器在各自的时间加上两个时间函数来获得复数倒谱。

公开(公告)号：US4914749A

公开(公告)日：1990-04-03

申请号：US665852

申请日：1984-10-29

申请人： Yukio Mitome

发明人： Yukio Mitome

IPC分类号： G10L11/00 ,  G10L19/02

CPC分类号： G10L19/02

摘要： A logarithmic frequency spectrum related to an input signal is converted by the use of an inverse Fourier transform into a cepstrum. The cepstrum has a first and a second frequency component which have a first peak and a second peak spaced apart from the first peak by a preselected period on an axis of frequency, respectively. The second frequency component is processed into a peak controlled frequency component having a controlled peak coincident with the first peak. The peak controlled frequency component and the first frequency component are summed up to produce an ultimate frequency component which corresponds to the value of the envelope parameter.

摘要翻译： 与输入信号相关的对数频谱通过使用逆傅里叶变换转换成倒谱。 倒谱具有第一和第二频率分量，它们分别具有第一峰值和与第一峰值间隔开频率轴上的预选周期的第二峰值。 第二频率分量被处理成具有与第一峰值一致的受控峰值的峰值受控频率分量。 将峰值受控频率分量和第一频率分量相加，以产生对应于包络参数值的最终频率分量。

公开(公告)号：US5892836A

公开(公告)日：1999-04-06

申请号：US738556

申请日：1996-10-28

申请人： Ryuuichi Ishige ,  Reishi Kondo ,  Yukio Mitome

发明人： Ryuuichi Ishige ,  Reishi Kondo ,  Yukio Mitome

IPC分类号： H03H15/00 ,  H03H17/02 ,  H03H21/00 ,  H04R25/00

CPC分类号： H04R25/505 ,  H04R2430/03 ,  H04R25/70

摘要： A digital hearing aid having a variable hearing compensating characteristics, comprises a hearing compensating circuit having a transposed transversal filter, an analyzer for frequency-analyzing an input signal, a memory storing a hearing characteristics of a person to be fitted with the hearing aid, and a controller receiving a frequency analysis result of the input signal and the hearing characteristics, for deriving coefficients for the transposed transversal filter to supply the derived coefficients to the transposed transversal filter. Since the transposed transversal filter is used, the S/N ration is improved, and the control of the characteristics of the filter becomes easy.

摘要翻译： 具有可变听力补偿特性的数字助听器包括具有转置横向滤波器的听力补偿电路，用于对输入信号进行频率分析的分析器，存储要配备助听器的人的听觉特性的存储器，以及 接收输入信号和听觉特性的频率分析结果的控制器，用于导出转置的横向滤波器的系数，以将导出的系数提供给转置的横向滤波器。 由于使用了转置的横向滤波器，所以提高了S / N比，并且滤波器特性的控制变得容易。

公开(公告)号：US5838801A

公开(公告)日：1998-11-17

申请号：US987617

申请日：1997-12-09

申请人： Ryuuichi Ishige ,  Yukio Mitome

发明人： Ryuuichi Ishige ,  Yukio Mitome

IPC分类号： H04R25/00

CPC分类号： H04R25/70 ,  H04R25/505

摘要： A digital hearing aid has input means for converting an input sound into a digital data for generating an input data, analyzing means for analyzing the input data converted by the input means by a digital conversion and calculating an acoustic pressure at each frequency band, control means for inputting a result of calculation by the analyzing means, acoustic sense characteristics storage means for preliminarily storing acoustic sense characteristics of a deafness and a person having healthy acoustic sense from a fitting means, gain calculation data storage means for preliminarily storing an acoustic pressure range the easiest to hear for the deafness from the fitting means, and acoustic sense compensating means for performing acoustic sense compensation process by amplifying the input data with a given gain. The control means calculates the gain of each frequency range on the basis of the acoustic sense characteristics and an acoustic pressure range stored in the acoustic sense characteristics storage means and the gain calculation data storage means.

摘要翻译： 数字助听器具有用于将输入声音转换为用于产生输入数据的数字数据的输入装置，用于通过数字转换分析由输入装置转换的输入数据并计算每个频带的声压的分析装置，控制装置 用于输入分析装置的计算结果的声学检测特性存储装置，用于从拟合装置预先存储耳聋的声学感觉特性和具有健康的声学感觉的人;增益计算数据存储装置，用于预先存储声压范围 通过拟合装置最容易听到耳聋的声学检测补偿装置，以及通过用给定增益放大输入数据来执行声学感测补偿处理的声学检测补偿装置。 控制装置根据存储在声感特性存储装置和增益计算数据存储装置中的声学感应特性和声压范围来计算每个频率范围的增益。

公开(公告)号：US5204905A

公开(公告)日：1993-04-20

申请号：US529421

申请日：1990-05-29

申请人： Yukio Mitome

发明人： Yukio Mitome

IPC分类号： G10L13/06 ,  G01L5/00 ,  G01L5/04 ,  G10L13/02 ,  G10L13/08

CPC分类号： G10L13/02

摘要： A text-to-speech synthesizer comprises an analyzer that decomposes a sequence of input characters into phoneme components and classifies them as a first group of phoneme components or a second group if they are to be synthesized by a speech parameter or by a formant rule, respectively. Speech parameters derived from natural human speech are stored in first memory locations corresponding to the phoneme components of the first group and the stored speech parameters are recalled from the first memory in response to each of the phoneme components of the first group. Formant rules capable of generating formant transition patterns are stored in second memory locations corresponding to the phoneme components of the second group, the formant rules being recalled from the second memory in response to each of the phoneme components of the second group. Formant transition patterns are derived from the formant rule recalled from the second memory, and formants of the derived transition patterns are converted into corresponding speech parameters. Spoken words are digitally synthesized from the speech parameters recalled from the first memory as well as from those supplied from the converted speech parameters.

摘要翻译： 文本到语音合成器包括分析器，其将输入字符的序列分解为音素分量，并将它们分类为第一组音素分量或第二​​组，如果它们要由语音参数或共振峰规则合成， 分别。 从自然人类语音导出的语音参数被存储在对应于第一组的音素分量的第一存储器位置中，并且响应于第一组的每个音素分量从第一存储器调用所存储的语音参数。 能够产生共振峰转换模式的共同规则被存储在对应于第二组的音素分量的第二存储器位置中，响应于第二组的每个音素分量，从第二存储器调用共振峰规则。 共同转变模式是从第二个存储器回收的共振峰规则中得到的，派生过渡模式的共同点被转换为相应的语音参数。 口语从根据从第一存储器召回的语音参数以及从转换的语音参数提供的语音参数进行数字合成。

公开(公告)号：US5835610A

公开(公告)日：1998-11-10

申请号：US774069

申请日：1996-12-23

申请人： Ryuuichi Ishige ,  Reishi Kondo ,  Yukio Mitome

发明人： Ryuuichi Ishige ,  Reishi Kondo ,  Yukio Mitome

IPC分类号： H02J1/00 ,  H04R25/00

CPC分类号： H04R25/43 ,  H04R25/554 ,  H04R25/505

摘要： An external device to be used along with a hearing aid device comprises an input means through which voice data are inputted, a hearing aid processor coupled to the input means for receiving the voice data from the input means to make an acoustic sense compensation of the voice data, and a transmitter coupled to the hearing aid processor for receiving compensated voice data from the hearing aid processor and transmitting the compensated voice data to the hearing aid device.

摘要翻译： 与助听器装置一起使用的外部设备包括输入语音数据的输入装置，耦合到输入装置的助听器处理器，用于从输入装置接收语音数据以对声音进行声学感测补偿 数据和耦合到助听器处理器的发射器，用于从助听器处理器接收经补偿的语音数据，并将补偿的话音数据发送到助听器装置。

公开(公告)号：US4845754A

公开(公告)日：1989-07-04

申请号：US010597

申请日：1987-02-03

申请人： Yukio Mitome

发明人： Yukio Mitome

IPC分类号： G10L11/00

CPC分类号： G10L25/00

摘要： A pole-zero analyzer includes an autocorrelation impulse response calculator, first and second buffers, a pole parameter value table, a first coefficient calculator, a first inverse filter, a second coefficient calculator, a second inverse filter, a pole-zero calculator for calculating a pole parameter and a zero parameter in response to outputs from the first and second inverse filters and calculating an error value for determining the pole parameter, a buffer memory for generating an output from the pole parameter value table obtained when the error value satisfies predetermined conditions and predetermined portions of the zero parameter and the pole parameter calculated by the pole-zero calculator as outputs of the pole-zero analyzer, and a control circuit for controlling operation timings of the respective parts.

摘要翻译： 极零分析器包括自相关脉冲响应计算器，第一和第二缓冲器，极参数值表，第一系数计算器，第一逆滤波器，第二系数计算器，第二逆滤波器，用于计算的极零计算器 响应于来自第一和第二反相滤波器的输出的极参数和零参数，并计算用于确定极参数的误差值;缓冲存储器，用于当误差值满足预定条件时，从所述极参数值表中产生输出 以及由零极点计算器计算的零参数和极点参数的预定部分作为极点分析器的输出，以及用于控制各个部分的操作定时的控制电路。

公开(公告)号：US4959866A

公开(公告)日：1990-09-25

申请号：US291827

申请日：1988-12-29

申请人： Noriko Matsuo ,  Yukio Mitome

发明人： Noriko Matsuo ,  Yukio Mitome

IPC分类号： G10L19/00 ,  G10L13/00 ,  G10L13/02 ,  G10L13/04

CPC分类号： G10L13/02

摘要： In order to simplify a speech synthesizer arrangement concurrently with improvement of operation flexibility thereof, a digital memory is arranged to store at least one voiced sound source and at least one unvoiced sound source. One of the sound sources is selected in accordance with the content of a first register, while the data within the selected source is specified by the content of a shift register sequence generator. Each of the bit patterns obtained at the shift register sequence generator is compared with the content of a second register. In the event that the contents of the sequence generator and the second register coincide, the shift register sequence generator is reset and/or the operating conditions(s) of the synthesizer is changed.

摘要翻译： 为了与其操作灵活性的改进同时简化语音合成器装置，数字存储器被布置成存储至少一个有声声源和至少一个清音声源。 根据第一寄存器的内容选择声源中的一个，而所选择的源中的数据由移位寄存器序列发生器的内容指定。 将在移位寄存器序列生成器处获得的每个位模式与第二寄存器的内容进行比较。 在序列发生器和第二寄存器的内容一致的情况下，移位寄存器序列生成器被复位和/或合成器的工作条件改变。

公开(公告)号：US6094489A

公开(公告)日：2000-07-25

申请号：US929771

申请日：1997-09-15

申请人： Ryuuichi Ishige ,  Yukio Mitome

发明人： Ryuuichi Ishige ,  Yukio Mitome

IPC分类号： H04R25/00

CPC分类号： H04R25/505 ,  H04R2225/43 ,  H04R25/356

摘要： Input data are analyzed by FFT, etc. in an analyzing section and power every frequency band is calculated and sent to a control section. In a gain control section, changing characteristics of a gain used in the control section are calculated on the basis of hearing ability characteristics of a user obtained from a memory section and a gain setting memory section, a sound pressure for starting a reduction in gain, and a sound pressure for setting the gain to be equal to or greater than 0 dB. The calculated changing characteristics are sent to the control section. In the control section, the gain every frequency band required in a hearing sense compensating section is determined on the basis of analyzed results obtained from the analyzing section, the hearing ability characteristics of the user obtained from the memory section, and the changing characteristics of the gain obtained from the gain control section. The control section sends data of the gain to the hearing sense compensating section. The hearing sense compensating section obtaining the input data and the gain data performs hearing sense compensation processing with respect to the input data and sends the processed input data to an output section.

摘要翻译： 在分析部分通过FFT等分析输入数据，并计算每个频带的功率并将其发送到控制部分。 在增益控制部中，基于从存储部和增益设定存储部获得的用户的听力能力特性，增益开始的声压， 以及用于将增益设定为等于或大于0dB的声压。 计算的变化特征被发送到控制部分。 在控制部分中，基于从分析部分获得的分析结果，从存储部分获得的用户的听力能力特性以及从存储部分获得的改变特性来确定听觉补偿部分中所需的每个频带的增益 从增益控制部获得的增益。 控制部将增益的数据发送给听觉补偿部。 获取输入数据和增益数据的听觉检测补偿部分对输入数据执行听觉补偿处理，并将处理的输入数据发送到输出部分。