专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

1. 发明申请

US20080120100A1 Method For Detecting Target Sound, Method For Detecting Delay Time In Signal Input, And Sound Signal Processor 审中-公开
标题翻译：用于检测目标声音的方法，用于检测信号输入和声音信号处理器中的延迟时间的方法
公开(公告)号：US20080120100A1
公开(公告)日：2008-05-22
申请号：US10509520
申请日：2004-03-17
申请人： Kazuya Takeda , Kiyoshi Tatara , Fumitada Itakura
发明人： Kazuya Takeda , Kiyoshi Tatara , Fumitada Itakura
IPC分类号： G10L21/02
CPC分类号： G10L15/20 , G10L21/0272 , G10L2021/02166
摘要： A sound signal processor includes first and second framing sections, first and second frequency analyzing sections, and a cross-spectrum calculating sections, for detecting the phase of a cross-spectrum between the sound signals input to first and second microphones, a phase extracting sections, a phase unwrap processing sections, a frequency band dividing section, and first through N-th inclination calculating sections, for detecting the inclinations of the phase of the cross-spectrum detected by the cross-spectrum calculating section with respect to the frequency, and a histogram calculating section and a voiced/voiceless determining section, for detecting a speech section in the sound received by the first and second microphones based on the inclination with respect to the frequency detected by the first through N-th inclination calculating sections.
摘要翻译：声音信号处理器包括第一和第二成帧部分，第一和第二频率分析部分以及用于检测输入到第一和第二麦克风的声音信号之间的交叉频谱的相位的交叉频谱计算部分，相位提取部分相位解除处理部，频带分割部以及第1〜第N倾斜计算部，检测交叉谱计算部检测出的交叉频谱的相位相对于频率的倾斜度，以及直方图计算部分和有声/无声确定部分，用于基于由第一至第N倾斜计算部分检测到的频率的倾斜度来检测由第一和第二麦克风接收的声音中的语音部分。

2. 发明授权

US06879968B1 SPEAKER VERIFICATION APPARATUS AND METHOD UTILIZING VOICE INFORMATION OF A REGISTERED SPEAKER WITH EXTRACTED FEATURE PARAMETER AND CALCULATED VERIFICATION DISTANCE TO DETERMINE A MATCH OF AN INPUT VOICE WITH THAT OF A REGISTERED SPEAKER 失效
标题翻译：扬声器验证装置和方法使用提取的特征参数的注册扬声器的语音信息和计算的验证距离来确定与注册的扬声器的输入语音的匹配
公开(公告)号：US06879968B1
公开(公告)日：2005-04-12
申请号：US09487467
申请日：2000-01-19
申请人： Shoji Hayakawa , Fumitada Itakura , Kazuya Takeda
发明人： Shoji Hayakawa , Fumitada Itakura , Kazuya Takeda
IPC分类号： G10L17/00 , G06F15/18 , G10L15/06 , G10L15/10 , G10L15/24
CPC分类号： G10L17/08
摘要： A speaker verification apparatus and method input an identity claim, select voice information of a registered speaker corresponding to the input identity claim, input a voice of the speaker, analyze the input voice so as to extract a feature parameter, calculate a verification distance between a feature parameter of the input voice and that of the voice of the registered speaker and the speaker distances between a feature parameter of the input voice and those of the voices of other speakers; and determine whether or not the input voice matches that of the registered speaker. A false acceptance error rate, is input as a threshold, and a probability distribution of interspeaker distances is functions to determine that the input voice is the voice of the registered person specified by the identity claim.
摘要翻译：扬声器验证装置和方法输入身份声明，选择与输入的身份声明相对应的已登记的说话者的语音信息，输入说话人的语音，分析输入的语音以便提取特征参数，计算一个输入声音的特征参数和已注册的扬声器的声音的特征参数以及输入声音的特征参数与其他扬声器的声音之间的扬声器距离; 并且确定输入的声音是否与所登记的扬声器的声音相匹配。误差接收误差率被输入为阈值，并且扬声器距离的概率分布是确定输入声音是由身份声明指定的注册人的声音的功能。

3. 发明授权

US4393272A Sound synthesizer 失效
标题翻译：声音合成器
公开(公告)号：US4393272A
公开(公告)日：1983-07-12
申请号：US188782
申请日：1980-09-19
申请人： Fumitada Itakura , Noboru Sugamura
发明人： Fumitada Itakura , Noboru Sugamura
IPC分类号： G10L21/04 , G10L13/02 , G10L19/06 , G10L1/00
CPC分类号： G10L19/07
摘要： The spectral envelope of speech signals are characterized by LSP (line spectral pair) parameters, termed w.sub.i and .theta..sub.i. In a speech synthesizer the control parameters for the filter may be a.sub.i =-2 cos w.sub.i and b.sub.i =-2 cos .theta..sub.i. The synthesizer filter is essentially an all pole filter consisting of two feedback paths each of which has a transfer function having zeroes on a unit circle in the complex plane.
摘要翻译：语音信号的频谱包络由LSP（线谱对）参数表征，称为wi和θi。在语音合成器中，滤波器的控制参数可以是ai = -2 cos wi和bi = -2cosθi。合成器滤波器基本上是由两个反馈路径组成的全极滤波器，每个反馈路径具有在复平面中的单位圆上具有零的传递函数。

4. 发明授权

US06934650B2 Noise signal analysis apparatus, noise signal synthesis apparatus, noise signal analysis method and noise signal synthesis method 失效
标题翻译：噪声信号分析装置，噪声信号合成装置，噪声信号分析方法和噪声信号合成方法
公开(公告)号：US06934650B2
公开(公告)日：2005-08-23
申请号：US10129076
申请日：2001-09-04
申请人： Koji Yoshida , Fumitada Itakura
发明人： Koji Yoshida , Fumitada Itakura
IPC分类号： G10L11/00 , G10L13/00 , G10L19/00 , H03M7/30 , G01L21/02
CPC分类号： G10L19/012 , G10L25/48
摘要： FFT section 102 transforms a windowed input noise signal into a frequency spectrum. Spectral model storing section 103 stores model information on spectral models. Spectral model series calculating section 104 calculates spectral model number series corresponding to amplitude spectral series of the input noise signal, using the model information stored in spectral model storing section 103. Duration model/transition probability calculating section 105 outputs model parameters using the spectral model number series calculated in spectral model series calculating section 104. It is thereby possible to synthesize a background noise with perceptual high quality.
摘要翻译： FFT部102将窗口化的输入噪声信号变换为频谱。光谱模型存储部分103存储关于光谱模型的模型信息。频谱模型系列计算部分104使用存储在频谱模型存储部分103中的模型信息来计算对应于输入噪声信号的幅度谱系列的频谱模型序列序列。持续时间模型/转移概率计算部分105使用在频谱模型系列计算部分104中计算的频谱模型序列序列输出模型参数。从而可以合成具有感知高品质的背景噪声。

5. 发明授权

US4538234A Adaptive predictive processing system 失效
标题翻译：自适应预测处理系统
公开(公告)号：US4538234A
公开(公告)日：1985-08-27
申请号：US414266
申请日：1982-09-02
申请人： Masaaki Honda , Fumitada Itakura , Nobuhiko Kitawaki
发明人： Masaaki Honda , Fumitada Itakura , Nobuhiko Kitawaki
IPC分类号： H03M7/32 , G10L19/02 , G10L19/04 , H03M7/30 , H04B1/66 , H04B14/06 , G10L1/00
CPC分类号： H04B1/667 , G10L19/04
摘要： An adaptive predictive coding (APC) and adaptive kit allocation (ABA) system which splits a speech signal into a plurality of bands, subjecting each signal to adaptive predictive coding and adaptively changes quantization characteristics in the adaptive predictive coding so that quantization noise may become small. An average amplitude (not average power) of the signal of each band is detected for each temporal sub-interval, and a quantization bit number and a quantization step size are adaptively determined from the average amplitude for each sub-interval, thereby permitting reduction of the quantity of hardware used.
摘要翻译：一种自适应预测编码（APC）和自适应套件分配（ABA）系统，其将语音信号分解成多个频带，对每个信号进行自适应预测编码，并自适应地改变自适应预测编码中的量化特性，使得量化噪声可能变小。对于每个时间子间隔检测每个频带的信号的平均幅度（不是平均功率），并且从每个子间隔的平均幅度自适应地确定量化位数和量化步长，从而允许使用的硬件数量。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式