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1. 发明申请

US20130063539A1 SIGNAL PROCESSING APPARATUS, SIGNAL PROCESSING METHOD, AND PROGRAM 审中-公开
标题翻译：信号处理装置，信号处理方法和程序
公开(公告)号：US20130063539A1
公开(公告)日：2013-03-14
申请号：US13587042
申请日：2012-08-16
申请人： Yohei SAKURABA , Yasuhiko Kato , Nobuyuki Kihara , Takeshi Yamaguchi
发明人： Yohei SAKURABA , Yasuhiko Kato , Nobuyuki Kihara , Takeshi Yamaguchi
IPC分类号： H04N7/14
CPC分类号： H04N7/142 , H04N5/232 , H04N5/262 , H04R3/005 , H04R5/04 , H04R27/00 , H04R2227/003 , H04R2430/20 , H04R2499/11
摘要： A signal processing apparatus includes: an audio separator that separates audios into a first audio and a second audio using two inputted audio signals; an audio combiner that combines the first audio with the second audio based on proportions of the audios separated by the audio separator; and an image combiner that combines a first image corresponding to the first audio with a second image corresponding to the second audio based on the proportions of the audios separated by the audio separator.
摘要翻译：信号处理装置包括：音频分离器，其使用两个输入的音频信号将音频分离成第一音频和第二音频; 音频组合器，其基于由音频分离器分离的音频的比例来组合第一音频和第二音频; 以及图像组合器，其基于由音频分离器分离的音频的比例，将对应于第一音频的第一图像与对应于第二音频的第二图像组合。

2. 发明申请

US20140178049A1 IMAGE PROCESSING APPARATUS, IMAGE PROCESSING METHOD, AND PROGRAM 审中-公开
标题翻译：图像处理设备，图像处理方法和程序
公开(公告)号：US20140178049A1
公开(公告)日：2014-06-26
申请号：US13823177
申请日：2012-08-01
申请人： Nobuyuki Kihara , Yohei Sakuraba , Takeshi Yamaguchi , Yasuhiko Kato
发明人： Nobuyuki Kihara , Yohei Sakuraba , Takeshi Yamaguchi , Yasuhiko Kato
IPC分类号： G11B27/036
CPC分类号： H04N5/262 , G03B31/00 , G10L15/26 , G10L2015/088 , H04N5/91 , H04N9/74 , H04N9/8211
摘要： This technology relates to an image processing apparatus, an image processing method, and a program, which enable easier addition of an effect to a moving image.In a portable terminal device, an ambient environmental sound and a voice uttered by a user are picked up by different sound pickup units when the moving image is shot. A keyword detecting unit detects a keyword determined in advance from the voice uttered by the user and an effect generating unit generates an image effect and a sound effect associated with the detected keyword. Then, an effect adding unit superposes the generated image effect on the shot moving image and synthesizes the generated sound effect with the environmental sound, thereby applying image effects and sound effects to the moving image. According to the portable terminal device, it is possible to easily add a desired effect to the moving image only by uttering the keyword while shooting the moving image. This technology may be applied to a mobile phone.
摘要翻译：该技术涉及一种图像处理装置，图像处理方法和程序，其使得能够更容易地向运动图像添加效果。在便携式终端装置中，当拍摄运动图像时，由不同的声音拾取单元拾取由用户发出的环境环境声音和语音。关键词检测单元从用户发出的语音中检测出预先确定的关键字，效果生成单元生成与检测到的关键字相关联的图像效果和声音效果。然后，效果添加单元将所产生的图像效果叠加在拍摄的运动图像上，并且利用环境声音合成生成的声音效果，从而将运动图像应用图像效果和声音效果。根据便携式终端装置，仅通过在拍摄运动图像时发出关键字，可以容易地向运动图像添加期望的效果。该技术可以应用于移动电话。

3. 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。

4. 发明授权

US08107617B2 Audio processor 失效
标题翻译：音频处理器
公开(公告)号：US08107617B2
公开(公告)日：2012-01-31
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： 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。

5. 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。

6. 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。

7. 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。

8. 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。

9. 发明授权

US07991166B2 Microphone apparatus 失效
标题翻译：麦克风设备
公开(公告)号：US07991166B2
公开(公告)日：2011-08-02
申请号：US11353088
申请日：2006-02-14
申请人： Nobuyuki Kihara , Yoshikazu Takahashi , Yasuhiko Kato
发明人： Nobuyuki Kihara , Yoshikazu Takahashi , Yasuhiko Kato
IPC分类号： H04R3/00 , H04R1/02 , G10L11/00 , G10L19/00
CPC分类号： H04R1/406 , H04R19/016 , H04R2201/401 , H04R2430/20
摘要： A microphone apparatus for processing and outputting an output signal of a microphone array including at least nine microphones includes a directivity function processing circuit that converts the output signal of the microphone array into a unidirectional signal and that outputs the unidirectional signal. The directivity function processing circuit expands a directivity function whose variable is an incident angle of an acoustic wave into a Fourier series up to at least third order. The variable in the expanded expression is produced from output signals of the microphones forming the microphone array.
摘要翻译：用于处理和输出包括至少九个麦克风的麦克风阵列的输出信号的麦克风装置包括将麦克风阵列的输出信号转换为单向信号并输出单向信号的方向性功能处理电路。指向性函数处理电路将可变为声波的入射角的方向性函数扩展为至少三阶的傅里叶级数。扩展表达式中的变量由形成麦克风阵列的麦克风的输出信号产生。

10. 发明申请

US20060188111A1 Microphone apparatus 失效
标题翻译：麦克风设备
公开(公告)号：US20060188111A1
公开(公告)日：2006-08-24
申请号：US11353088
申请日：2006-02-14
申请人： Nobuyuki Kihara , Yoshikazu Takahashi , Yasuhiko Kato
发明人： Nobuyuki Kihara , Yoshikazu Takahashi , Yasuhiko Kato
IPC分类号： H04R3/00 , H04R1/02 , H04R9/08
CPC分类号： H04R1/406 , H04R19/016 , H04R2201/401 , H04R2430/20
摘要： A microphone apparatus for processing and outputting an output signal of a microphone array including at least nine microphones includes a directivity function processing circuit that converts the output signal of the microphone array into a unidirectional signal and that outputs the unidirectional signal. The directivity function processing circuit expands a directivity function whose variable is an incident angle of an acoustic wave into a Fourier series up to at least third order. The variable in the expanded expression is produced from output signals of the microphones forming the microphone array.
摘要翻译：用于处理和输出包括至少九个麦克风的麦克风阵列的输出信号的麦克风装置包括将麦克风阵列的输出信号转换为单向信号并输出单向信号的方向性功能处理电路。指向性函数处理电路将可变为声波的入射角的方向性函数扩展为至少三阶的傅里叶级数。扩展表达式中的变量由形成麦克风阵列的麦克风的输出信号产生。

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