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1. 发明授权

US08098843B2 Sound source direction detecting apparatus, sound source direction detecting method, and sound source direction detecting camera 失效
标题翻译：声源方向检测装置，声源方向检测方法和声源方向检测摄像机
公开(公告)号：US08098843B2
公开(公告)日：2012-01-17
申请号：US12284455
申请日：2008-09-22
申请人： Takayoshi Kawaguchi , Yasuhiro Kodama , Yohei Sakuraba
发明人： Takayoshi Kawaguchi , Yasuhiro Kodama , Yohei Sakuraba
IPC分类号： H04R3/00 , H04N7/00
CPC分类号： G01S3/801 , G01S3/8083 , G01S3/86
摘要： Disclosed herein is a sound source direction detecting apparatus including: a plurality of microphones configured to collect sounds from a sound source in order to form an audio frame; a frequency decomposition section configured to decompose the audio frame into frequency components; an error range determination section configured to determine the effects of noises collected together with the sounds as an error range relative to phases; a power level dispersion section configured to disperse power levels of the sounds for each of the frequency components decomposed by the frequency decomposition section, on the basis of the error range determined by the error range determination section; a power level addition section configured to add the power levels dispersed by the power level dispersion section; and a sound source direction detection section configured to detect the direction of the sound source based on the phase at which is located the highest of the power levels added by the power level addition section.
摘要翻译：这里公开了一种声源方向检测装置，包括：多个麦克风，被配置为从声源收集声音以形成音频帧; 频率分解部，被配置为将音频帧分解成频率分量; 误差范围确定部分，被配置为将与所述声音一起收集的噪声的影响确定为相对于相位的误差范围; 功率电平分散部，其基于由所述误差范围确定部确定的误差范围，分配由所述频率分解部分解的每个频率分量的声音的功率电平; 功率电平相加部分，被配置为添加由功率电平分散部分分散的功率电平; 以及声源方向检测部，被配置为基于位于功率电平相加部附加的最高功率电平的相位来检测声源的方向。

2. 发明申请

US20090086993A1 Sound source direction detecting apparatus, sound source direction detecting method, and sound source direction detecting camera 失效
标题翻译：声源方向检测装置，声源方向检测方法和声源方向检测摄像机
公开(公告)号：US20090086993A1
公开(公告)日：2009-04-02
申请号：US12284455
申请日：2008-09-22
申请人： Takayoshi Kawaguchi , Yasuhiro Kodama , Yohei Sakuraba
发明人： Takayoshi Kawaguchi , Yasuhiro Kodama , Yohei Sakuraba
IPC分类号： H04R3/00 , H04N7/14
CPC分类号： G01S3/801 , G01S3/8083 , G01S3/86
摘要： Disclosed herein is a sound source direction detecting apparatus including: a plurality of microphones configured to collect sounds from a sound source in order to form an audio frame; a frequency decomposition section configured to decompose the audio frame into frequency components; an error range determination section configured to determine the effects of noises collected together with the sounds as an error range relative to phases; a power level dispersion section configured to disperse power levels of the sounds for each of the frequency components decomposed by the frequency decomposition section, on the basis of the error range determined by the error range determination section; a power level addition section configured to add the power levels dispersed by the power level dispersion section; and a sound source direction detection section configured to detect the direction of the sound source based on the phase at which is located the highest of the power levels added by the power level addition section.
摘要翻译：这里公开了一种声源方向检测装置，包括：多个麦克风，被配置为从声源收集声音以形成音频帧; 频率分解部，被配置为将音频帧分解成频率分量; 误差范围确定部分，被配置为将与所述声音一起收集的噪声的影响确定为相对于相位的误差范围; 功率电平分散部，其基于由所述误差范围确定部确定的误差范围，分配由所述频率分解部分解的每个频率分量的声音的功率电平; 功率电平相加部分，被配置为添加由功率电平分散部分分散的功率电平; 以及声源方向检测部，被配置为基于位于功率电平相加部附加的最高功率电平的相位来检测声源的方向。

3. 发明授权

US08160239B2 Echo canceller and speech processing apparatus 失效
标题翻译：回波消除器和语音处理装置
公开(公告)号：US08160239B2
公开(公告)日：2012-04-17
申请号：US11872936
申请日：2007-10-16
申请人： Yohei Sakuraba , Nobuyuki Kihara , Takayoshi Kawaguchi
发明人： Yohei Sakuraba , Nobuyuki Kihara , Takayoshi Kawaguchi
IPC分类号： H04M9/08
CPC分类号： H04M9/082
摘要： An echo canceller used for hands-free communication systems in which hands-free communication is performed by using a speaker and a microphone is disclosed. The echo canceller includes a step size control unit calculating a step size value in an adaptive filter and an adaptive filter unit estimating an echo component of a feedback path from an input signal to the feedback path by adaptively identifying an impulse response of the feedback path formed by an acoustical coupling and the like of the speaker and the microphone, and subtracting the echo component from an output signal from the feedback path, in which the step size control unit calculates a step size value by using an echo reduction amount defined based on the ratio between the output signal from the feedback path and a residual signal and outputs the value to the adaptive filter unit.
摘要翻译：公开了一种用于通过使用扬声器和麦克风进行免提通信的免提通信系统的回声消除器。回波消除器包括步进大小控制单元，其计算自适应滤波器中的步长值，以及自适应滤波器单元，通过自适应地识别形成的反馈路径的脉冲响应来估计从输入信号到反馈路径的反馈路径的回波分量通过扬声器和麦克风的声耦合等，以及从反馈路径的输出信号中减去回波分量，其中步长控制单元通过使用基于所述反馈路径定义的回波减少量来计算步长值来自反馈路径的输出信号与残差信号之间的比值，并将该值输出到自适应滤波器单元。

4. 发明授权

US08913737B2 Echo canceller and microphone apparatus 有权
标题翻译：回声消除器和麦克风设备
公开(公告)号：US08913737B2
公开(公告)日：2014-12-16
申请号：US11505115
申请日：2006-08-16
申请人： Takayoshi Kawaguchi , Yohei Sakuraba
发明人： Takayoshi Kawaguchi , Yohei Sakuraba
IPC分类号： H04M9/08
CPC分类号： H04M9/082
摘要： An echo canceller for executing adaptive processing for canceling an echo component mixed with an audio input signal includes a volume ratio learner configured to compute a volume ratio between an audio output signal externally outputted and the audio input signal mixed with an echo component caused by reflection of the audio output signal to the audio input signal, thereby learning the volume ratio in a regular status in own apparatus, a double-talk detector configured to detect the double-talk status depending on whether a this-time volume ratio computed this time adapts to a double-talk status predicted by the learning of volume ratio and an echo cancel processor configured to control a learning operation of the echo component for the adaptive processing on the basis of a result of the double-talk status detection by the double-talk detector.
摘要翻译：用于执行用于消除与音频输入信号混合的回波分量的自适应处理的回波消除器包括体积比学习器，其被配置为计算外部输出的音频输出信号与由回波分量混合的音频输入信号之间的体积比，音频输出信号输入到音频输入信号，从而在自己的装置中学习常规状态的体积比，双方通话检测器，被配置为根据本次计算的这个时间体积比是否适应于通过体积比学习预测的双方通话状态和回波消除处理器，其被配置为基于双方通话检测器的双方通话状态检测的结果来控制用于自适应处理的回波分量的学习操作。

5. 发明申请

US20070041576A1 Echo canceller and microphone apparatus 有权
标题翻译：回声消除器和麦克风设备
公开(公告)号：US20070041576A1
公开(公告)日：2007-02-22
申请号：US11505115
申请日：2006-08-16
申请人： Takayoshi Kawaguchi , Yohei Sakuraba
发明人： Takayoshi Kawaguchi , Yohei Sakuraba
IPC分类号： A61F11/06 , H04M9/08 , G10K11/16 , H03B29/00
CPC分类号： H04M9/082
摘要： An echo canceller for executing adaptive processing for canceling an echo component mixed with an audio input signal includes a volume ratio learner configured to compute a volume ratio between an audio output signal externally outputted and the audio input signal mixed with an echo component caused by reflection of the audio output signal to the audio input signal, thereby learning the volume ratio in a regular status in own apparatus, a double-talk detector configured to detect the double-talk status depending on whether a this-time volume ratio computed this time adapts to a double-talk status predicted by the learning of volume ratio and an echo cancel processor configured to control a learning operation of the echo component for the adaptive processing on the basis of a result of the double-talk status detection by the double-talk detector.
摘要翻译：用于执行用于消除与音频输入信号混合的回波分量的自适应处理的回波消除器包括体积比学习器，其被配置为计算外部输出的音频输出信号与由回波分量混合的音频输入信号之间的体积比，音频输出信号输入到音频输入信号，从而在自己的装置中学习常规状态的体积比，双方通话检测器，被配置为根据本次计算的这个时间体积比是否适应于通过体积比学习预测的双方通话状态和回波消除处理器，其被配置为基于双方通话检测器的双方通话状态检测的结果来控制用于自适应处理的回波分量的学习操作。

6. 发明申请

US20100183163A1 SOUND SIGNAL PROCESSOR AND DELAY TIME SETTING METHOD 审中-公开
标题翻译：声信号处理器和延迟时间设置方法
公开(公告)号：US20100183163A1
公开(公告)日：2010-07-22
申请号：US12663332
申请日：2008-06-05
申请人： Takeshi Matsui , Yasuhiko Kato , Nobuyuki Kihara , Hideki Kishi , Yasuhiro Kodama , Yohei Sakuraba
发明人： Takeshi Matsui , Yasuhiko Kato , Nobuyuki Kihara , Hideki Kishi , Yasuhiro Kodama , Yohei Sakuraba
IPC分类号： H04B3/20
CPC分类号： H04R3/02 , H04M9/082
摘要： An echo canceller formed of an adaptive filter is designed such that even under a condition where a system transmission delay is undefined, an appropriate delay time can be set in a delay circuit that absorbs a system delay, and that an effective echo cancellation effect can always be achieved. A time difference of a transmission path until a reproduction audio signal input to the delay circuit is input as a processing target signal of an adaptive filter system through a space between a speaker and a microphone is determined, and the delay time corresponding to this time difference is set in the delay circuit. At this time, the speaker and the microphone are placed so that the distance therebetween is small, and the delay time of the delay circuit is set to 0. Thus, the determined time difference indicates a system transmission delay in the above transmission path. That is, an accurate delay time corresponding to the system transmission delay can be set in the delay circuit.
摘要翻译：由自适应滤波器构成的回波消除器被设计成即使在系统传输延迟未定义的条件下，也可以在吸收系统延迟的延迟电路中设置适当的延迟时间，并且有效的回波消除效果可以始终实现。确定通过扬声器和麦克风之间的空间输入到延迟电路输入的再现音频信号之间的传输路径的时间差作为自适应滤波器系统的处理目标信号，并且确定对应于该时间差的延迟时间被设置在延迟电路中。此时，扬声器和麦克风被放置成使得它们之间的距离小，并且延迟电路的延迟时间被设置为0.因此，所确定的时间差表示上述传输路径中的系统传输延迟。也就是说，可以在延迟电路中设置与系统传输延迟相对应的准确的延迟时间。

7. 发明申请

US20090154692A1 Voice processing apparatus, voice processing system, and voice processing program 失效
标题翻译：语音处理装置，语音处理系统和语音处理程序
公开(公告)号：US20090154692A1
公开(公告)日：2009-06-18
申请号：US12316112
申请日：2008-12-09
申请人： Yohei Sakuraba , Yasuhiko Kato
发明人： Yohei Sakuraba , Yasuhiko Kato
IPC分类号： H04B3/20 , H04M9/08
CPC分类号： H04M9/082
摘要： A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.
摘要翻译：一种语音处理装置，包括将由第一麦克风产生的第一语音信号和由第二麦克风产生的第二语音信号划分成预定频带的频带划分部分，分离由第一麦克风发出的声音的回波分量的声源分离部分声源包括在基于第一和第二麦克风的功率的每个预定频带中的由第二声源发出的声音中的声源，以及合成第一和第二语音信号的频带合成部分，其中第一声源已经被声源隔离部分隔离成包括由第一声源发出的声音的声音信号和包括第一声源的回波分量的声音信号。

8. 发明申请

US20090150151A1 Audio processing apparatus, audio processing system, and audio processing program 审中-公开
标题翻译：音频处理装置，音频处理系统和音频处理程序
公开(公告)号：US20090150151A1
公开(公告)日：2009-06-11
申请号：US12313334
申请日：2008-11-19
申请人： Yohei Sakuraba , Yasuhiko Kato
发明人： Yohei Sakuraba , Yasuhiko Kato
IPC分类号： G10L17/00
CPC分类号： G10L21/028 , G10L17/00 , G10L21/04
摘要： Disclosed herein is an audio processing apparatus for processing a plurality of pieces of audio data of sounds picked up by a plurality of microphones. The apparatus includes: a speaker identification section configured to identify a speaker based on the audio data; a simultaneous speech section identification section configured to, when at least first and second speakers have been identified, identify speech sections during which the first and second speakers have made speeches, and identify a section during which the first and second speakers have made the speeches at the same time as a simultaneous speech section; and an arranging section configured to separate audio data of the first speaker and audio data of the second speaker from the simultaneous speech section, and allow the audio data of the first speaker and the audio data of the second speaker to be outputted at mutually different timings.
摘要翻译：这里公开了一种用于处理由多个麦克风拾取的声音的多个音频数据的音频处理装置。该装置包括：扬声器识别部，被配置为基于音频数据识别扬声器; 同时语音部分识别部分，被配置为当已经识别出至少第一和第二扬声器时，识别第一和第二扬声器已经进行演讲的语音部分，并且识别第一和第二扬声器在其中发出演讲的部分同时作为同声传译部分; 以及布置部，被配置为将第一扬声器的音频数据和第二扬声器的音频数据与同步语音部分分离，并且允许第一说话者的音频数据和第二说话者的音频数据以相互不同的定时输出。

9. 发明申请

US20070206817A1 AUDIO PROCESSOR 失效
标题翻译：音频处理器
公开(公告)号：US20070206817A1
公开(公告)日：2007-09-06
申请号：US11681025
申请日：2007-03-01
申请人： Yohei Sakuraba , Yasuhiko Kato , Nobuyuki Kihara
发明人： Yohei Sakuraba , Yasuhiko Kato , Nobuyuki Kihara
IPC分类号： H04B3/20 , H04M9/08
CPC分类号： H04M9/082
摘要： An audio processor of a loud speech communication system including a speaker and a microphone is provided. The audio processor includes: an adaptive filter wherein an amount of update in a learning event is set to an arbitrary value, and a filter coefficient is serially determined corresponding to the set amount of update; a semi-fixed filter adapted to an echo cancellation process of an audio input signal input from the microphone; adaptive filter assessment unit that calculates a length of an update vector based on the filter coefficient determined by the adaptive filter and a length of an update vector based on a filter coefficient set in the semi-fixed filter and that performs assessment of the filter coefficients in accordance with the update vectors; and coefficient specifying unit that sets an optimal filter coefficient among the filter coefficients into the semi-fixed filter in accordance with the result of the assessment of the filter coefficients performed by the adaptive filter assessment unit.
摘要翻译：提供了包括扬声器和麦克风的大声语音通信系统的音频处理器。音频处理器包括：自适应滤波器，其中将学习事件中的更新量设置为任意值，并且根据设定的更新量对滤波器系数进行串行确定; 适用于从麦克风输入的音频输入信号的回波消除处理的半固定滤波器; 自适应滤波器评估单元，其基于由所述自适应滤波器确定的滤波器系数和基于在所述半固定滤波器中设置的滤波器系数的更新向量的长度来计算更新向量的长度，并且执行所述滤波器系数的估计按照更新向量; 以及系数指定单元，其根据由自适应滤波器评估单元执行的滤波器系数的评估结果，将滤波器系数中的最佳滤波器系数设置为半固定滤波器。

10. 发明授权

US08194851B2 Voice processing apparatus, voice processing system, and voice processing program 失效
标题翻译：语音处理装置，语音处理系统和语音处理程序
公开(公告)号：US08194851B2
公开(公告)日：2012-06-05
申请号：US12316112
申请日：2008-12-09
申请人： Yohei Sakuraba , Yasuhiko Kato
发明人： Yohei Sakuraba , Yasuhiko Kato
IPC分类号： H04M9/08 , H04R29/00 , H04B15/00
CPC分类号： H04M9/082
摘要： A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.
摘要翻译：一种语音处理装置，包括将由第一麦克风产生的第一语音信号和由第二麦克风产生的第二语音信号划分成预定频带的频带划分部分，分离由第一麦克风发出的声音的回波分量的声源分离部分声源包括在基于第一和第二麦克风的功率的每个预定频带中的由第二声源发出的声音中的声源，以及合成第一和第二语音信号的频带合成部分，其中第一声源已经被声源隔离部分隔离成包括由第一声源发出的声音的声音信号和包括第一声源的回波分量的声音信号。

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