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31. 发明授权

US08473291B2 Sound processing apparatus, apparatus and method for controlling gain, and computer program 有权
标题翻译：用于控制增益的声音处理装置，装置和方法以及计算机程序
公开(公告)号：US08473291B2
公开(公告)日：2013-06-25
申请号：US12232177
申请日：2008-09-11
申请人： Naoshi Matsuo
发明人： Naoshi Matsuo
IPC分类号： G10L21/00
CPC分类号： H03G3/32
摘要： A sound processing apparatus is provided for estimating the power of background noise using a directional sound receiving technology using a plurality of sound receiving units, computing a gain control value on the basis of the estimated power of background noise and a predetermined power target value, and outputting the gain control value, so that a delay time of starting gain control can be reduced, and a slow response of a speech recognition application program or degradation of the speech quality of a voice communication program can be prevented.
摘要翻译：提供了一种使用使用多个声音接收单元的定向声音接收技术来估计背景噪声的功率的声音处理装置，基于背景噪声的估计功率和预定的功率目标值来计算增益控制值，以及输出增益控制值，使得可以减少开始增益控制的延迟时间，并且可以防止语音识别应用程序的缓慢响应或语音通信程序的语音质量的劣化。

32. 发明授权

US08340963B2 Echo suppressing system, echo suppressing method, recording medium, echo suppressor, sound output device, audio system, navigation system and mobile object 有权
标题翻译：回波抑制系统，回波抑制方法，记录介质，回波抑制器，声音输出装置，音响系统，导航系统和移动物体
公开(公告)号：US08340963B2
公开(公告)日：2012-12-25
申请号：US12756768
申请日：2010-04-08
申请人： Naoshi Matsuo , Taisuke Itou
发明人： Naoshi Matsuo , Taisuke Itou
IPC分类号： G10L21/02 , G10L15/00 , A61F11/06 , G10K11/16
CPC分类号： H04R3/02 , G10L15/20 , G10L2021/02082
摘要： An echo suppressing system includes: a sound output device for outputting sound based on a sound signal, including a passing section for allowing passage of a component of a different frequency band, and a plurality of sound output sections, each of which outputs sound based on each of the plurality of sound signals passed through the passing section; a summer for summing the plurality of sound signals to generate a reference sound signal; a sound input device for converting input sound into a sound signal; and an echo suppressor for suppressing echo based on the sound output by the sound output device, including an input section to which a sound signal is input from the sound input device as an observation sound signal, and a correction section for correcting the observation sound signal so as to suppress echo included in the observation sound signal.
摘要翻译：回波抑制系统包括：声音输出装置，用于输出基于声音信号的声音，包括用于允许不同频带的分量通过的通过部分和多个声音输出部分，每个声音输出部分基于多个声音信号中的每一个通过通过部分; 用于对所述多个声音信号求和以产生参考声音信号的加法器; 用于将输入声音转换成声音信号的声音输入装置; 以及基于声音输出装置输出的声音抑制回波的回波抑制器，包括从声音输入装置输入声音信号的输入部分作为观察声音信号，以及校正部分，用于校正观察声音信号以抑制观察声信号中包含的回波。

33. 发明授权

US08160264B2 Transfer function estimating device, noise suppressing apparatus and transfer function estimating method 有权
标题翻译：传递函数估计装置，噪声抑制装置和传递函数估计方法
公开(公告)号：US08160264B2
公开(公告)日：2012-04-17
申请号：US12453177
申请日：2009-04-30
申请人： Taisuke Itou , Naoshi Matsuo
发明人： Taisuke Itou , Naoshi Matsuo
IPC分类号： A61F11/06 , G10K11/16
CPC分类号： G10K11/178 , G10K2210/1282 , G10K2210/30232 , G10K2210/3033 , G10K2210/3055 , H04R2499/13 , H04S7/303
摘要： A transfer function estimating device for estimating a transfer function of a sound, includes: a sound receiving module receiving a sound from a given sound source and converting the sound into a tone signal; a storage module storing first transfer functions of the sound propagating from the given sound source to the sound receiving module and transformation coefficients for converting the first transfer functions into given second transfer functions so as to associate with each other; a reference tone signal acquiring module acquiring a reference tone signal of the sound source; an acquiring module acquiring a transfer function of the sound received by the sound receiving module on the basis of the tone signal and the reference tone signal; a specifying module acquiring a cross-correlation value between the transfer function acquired by the acquiring module and each of the first transfer functions stored in the storage module.
摘要翻译：用于估计声音的传递函数的传递函数估计装置包括：声音接收模块，接收来自给定声源的声音并将声音转换成音调信号; 存储模块，其将从所述给定声源传播的声音的第一传递函数存储到所述声音接收模块;以及变换系数，用于将所述第一传递函数转换为给定的第二传递函数，以便彼此关联; 参考音信号获取模块，用于获取所述声源的参考音信号; 获取模块，基于所述音调信号和所述参考音调信号，获取由所述声音接收模块接收的声音的传递函数; 指定模块获取由所述获取模块获取的所述传递函数与存储在所述存储模块中的所述第一传递函数之间的互相关值。

34. 发明授权

US08036888B2 Collecting sound device with directionality, collecting sound method with directionality and memory product 有权
标题翻译：收集具有方向性的声音装置，收集方向性和记忆产品的声音方法
公开(公告)号：US08036888B2
公开(公告)日：2011-10-11
申请号：US11519792
申请日：2006-09-13
申请人： Naoshi Matsuo
发明人： Naoshi Matsuo
IPC分类号： G10L15/20
CPC分类号： H04R1/406 , H04R3/04 , H04R2430/20
摘要： A sound input from sound sources existing in a plurality of directions is accepted and converted into a signal on a frequency axis. A suppressing function to suppress the converted signal on a frequency axis is computed, an amplitude component of a signal on a frequency axis is multiplied by the computed suppressing function and the converted signal on a frequency axis is corrected. A phase component of each converted signal on a frequency axis is computed for each frequency and a difference of phase components is computed. A probability value indicative of probability of existence of a sound source in a predetermined direction is specified based on the computed difference and a suppressing function to suppress a sound input from a sound source other than a sound source in a predetermined direction is computed based on the specified probability value.
摘要翻译：接收来自存在于多个方向的声源的声音输入并将其转换为频率轴上的信号。计算抑制频率轴上的转换信号的抑制功能，将频率轴上的信号的振幅分量乘以计算出的抑制功能，并对频率轴上的转换信号进行校正。针对每个频率计算频率轴上的每个转换信号的相位分量，并且计算相位分量的差。基于所计算出的差异来指定指示声源在预定方向上的概率存在的概率值，并且基于所计算的差异来计算抑制从预定方向上的声源以外的声源输入的声音的抑制功能指定概率值。

35. 发明授权

US07979278B2 Speech recognition system and speech file recording system 有权
标题翻译：语音识别系统和语音文件记录系统
公开(公告)号：US07979278B2
公开(公告)日：2011-07-12
申请号：US10285482
申请日：2002-11-01
申请人： Naoshi Matsuo
发明人： Naoshi Matsuo
IPC分类号： G10L15/04
CPC分类号： G10L15/07
摘要： A user term information extraction unit extracts term information of a user out of information that has been input by the user to an application for use other than speech recording beforehand, and a speech recognition dictionary management unit expands a vocabulary of a speech recognition dictionary according to the term information of the user. Next, the user inputs speech via a speech input unit, and a speech recognition unit executes speech recognition using the speech recognition dictionary. A representative term information selection unit extracts the term information of the user contained in the speech recognition result, and selects one or a plurality of pieces of representative term information from the term information of the user. A speech file recording unit records the speech data as a speech file, and renders a file name of the speech file according to the representative term information. Thus, a speech recognition and speech file recording system is provided that records input data as a speech file and automatically renders a file name thereto with which a content of the file can be grasped immediately.
摘要翻译：用户术语信息提取单元从事先输入的用户输入的信息中提取用户的术语信息，语音识别词典管理单元根据语音识别词典扩展语音识别词典的词汇用户的术语信息。接下来，用户经由语音输入单元输入语音，并且语音识别单元使用语音识别词典来执行语音识别。代表词语信息选择单元提取包含在语音识别结果中的用户的术语信息，并从用户的术语信息中选择一个或多个代表性词汇信息。语音文件记录单元将语音数据记录为语音文件，并根据代表性词汇信息呈现语音文件的文件名。因此，提供了将输入数据作为语音文件记录的语音识别和语音文件记录系统，并且自动呈现可以立即掌握文件内容的文件名。

36. 发明申请

US20100111325A1 Device for processing sound signal, and method of processing sound signal 有权
标题翻译：用于处理声音信号的装置，以及处理声音信号的方法
公开(公告)号：US20100111325A1
公开(公告)日：2010-05-06
申请号：US12458957
申请日：2009-07-28
申请人： Naoshi Matsuo
发明人： Naoshi Matsuo
IPC分类号： H04R3/00
CPC分类号： H04R3/005 , G01C21/3664
摘要： A signal processing device includes at least two sound signal input elements, an orthogonal transformer, a phase difference determiner, and a virtual spectrum generator. Two sound signals in a time domain are received from the two respective sound signal input elements. The orthogonal transformer orthogonally transforms the two sound signals in the time domain into two respective spectral signals in a frequency domain. The phase difference determiner determines a phase difference between the two spectral signals in the frequency domain. The virtual spectrum generator generates a virtual spectral signal in the frequency domain in accordance with the phase difference and one of the two spectral signals.
摘要翻译：信号处理装置包括至少两个声音信号输入元件，正交变换器，相位差确定器和虚拟频谱发生器。从两个相应的声音信号输入元件接收时域中的两个声音信号。正交变换器将时域中的两个声音信号正交地变换成频域中的两个相应的频谱信号。相位差确定器确定频域中的两个频谱信号之间的相位差。虚拟频谱发生器根据相位差和两个频谱信号之一产生频域中的虚拟频谱信号。

37. 发明申请

US20090010445A1 ECHO SUPPRESSOR, ECHO SUPPRESSING METHOD, AND COMPUTER READABLE STORAGE MEDIUM 有权
标题翻译： ECHO抑制器，ECHO抑制方法和计算机可读存储介质
公开(公告)号：US20090010445A1
公开(公告)日：2009-01-08
申请号：US12142826
申请日：2008-06-20
申请人： Naoshi Matsuo
发明人： Naoshi Matsuo
IPC分类号： H04B3/20
CPC分类号： H04M9/082
摘要： An apparatus is provided for suppressing an echo signal included in a measured signal corresponding to a measured sound. In the apparatus, the measured signal and a reference signal in a time domain are transformed into a frequency domain, and calculated for obtaining each value of a ratio and a correlation between the measured signal and the reference signal in the frequency domain. With executing a comparison of the values of the ratio and the correlation, a coefficient is derived, where a product of the coefficient and the measured sound in the frequency domain gives an estimated value of the echo signal. The echo in the measured signal is suppressed with subtracting the estimation of the echo signal from the measured signal, respectively in the frequency domain.
摘要翻译：提供了一种用于抑制包括在与测量声音相对应的测量信号中的回波信号的装置。在该装置中，将测量信号和时域中的参考信号变换为频域，并计算得到频域中的每个比值和测量信号与参考信号之间的相关性。通过执行比率值和相关性的比较，导出系数，其中系数和频域中的测量声音的乘积给出回波信号的估计值。在频域中分别从测量信号中减去回波信号的估计值来抑制测量信号中的回波。

38. 发明申请

US20080085012A1 Sound signal correcting method, sound signal correcting apparatus and computer program 有权
标题翻译：声音信号校正方法，声音信号校正装置和计算机程序
公开(公告)号：US20080085012A1
公开(公告)日：2008-04-10
申请号：US11698113
申请日：2007-01-26
申请人： Naoshi Matsuo
发明人： Naoshi Matsuo
IPC分类号： H04B15/00 , G10L15/20 , G10L15/00
CPC分类号： G10L21/0208
摘要： A sound signal correcting apparatus converts an acquired sound signal into a phase spectrum and an amplitude spectrum by an FFT process, compares the amplitude spectrum of the obtained sound signal with a noise model so that a correction coefficient used for correcting the amplitude spectrum of the sound signal is derived, smoothes waveform of the amplitude spectrum of the sound signal using the derived correction coefficient, and converts the sound signal into a sound signal where the amplitude spectrum is corrected by performing an inverse FFT process on the phase spectrum and the smoothed amplitude spectrum.
摘要翻译：声音信号校正装置通过FFT处理将所获取的声音信号转换为相位谱和幅度谱，将所获得的声音信号的振幅谱与噪声模型进行比较，以便用于校正声音的幅度谱的校正系数导出信号，使用导出的校正系数平滑声音信号的振幅谱的波形，并且通过对相位谱进行逆FFT处理并且将平滑的幅度谱转换成声音信号，该声音信号被校正了振幅谱。

39. 发明授权

US07072310B2 Echo canceling system 有权
公开(公告)号：US07072310B2
公开(公告)日：2006-07-04
申请号：US09867664
申请日：2001-05-31
申请人： Naoshi Matsuo
发明人： Naoshi Matsuo
IPC分类号： H04B3/20
CPC分类号： H04M9/082
摘要： An optimal echo canceling processing is provided regardless of a condition of a partner's system and even in the condition with large delay amount and large fluctuation amplitude of a network transmission. A speaker 10 inputs a voice via a microphone 11. A voice signal of this voice is transmitted to a terminal 14 of a conversation partner via a VoIP application 13 and the internet 30. Concurrently, the voice signal is inputted to an echo canceller 100. The echo canceller 100 detects sound characteristics of an echo path in advance or dynamically, adjusts a filter coefficient for generating an echo canceling signal, and receives an adjustment by a user. The echo canceller 100 generates the echo canceling signal by processing the received voice signal based on the sound characteristics and the adjusting amount. The echo canceling signal is subtracted from a response signal containing an echo that has been re-inputted from a loudspeaker 22 of the partner via a microphone 21.

40. 发明授权

US07035416B2 Microphone array apparatus 有权
公开(公告)号：US07035416B2
公开(公告)日：2006-04-25
申请号：US10003768
申请日：2001-10-26
申请人： Naoshi Matsuo
发明人： Naoshi Matsuo
IPC分类号： H04R3/00
CPC分类号： H04R3/005 , H04R2201/403 , H04R2410/05 , H04R2430/23
摘要： A microphone array apparatus includes a microphone array including microphones, one of the microphones being a reference microphone, filters receiving output signals of the microphones, and a filter coefficient calculator which receives the output signals of the microphones, a noise and a residual signal obtained by subtracting filtered output signals of the microphones other than the reference microphone from a filtered output signal of the reference microphone and which obtain filter coefficients of the filters in accordance with an evaluation function based on the residual signal.

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