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    • 1. 发明授权
    • Echo canceller and speech processing apparatus
    • 回波消除器和语音处理装置
    • US08160239B2
    • 2012-04-17
    • US11872936
    • 2007-10-16
    • Yohei SakurabaNobuyuki KiharaTakayoshi Kawaguchi
    • Yohei SakurabaNobuyuki KiharaTakayoshi Kawaguchi
    • H04M9/08
    • H04M9/082
    • 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.
    • 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。
    • 2. 发明申请
    • Echo Canceller and Speech Processing Apparatus
    • 回声消除器和语音处理装置
    • US20080260172A1
    • 2008-10-23
    • US11872936
    • 2007-10-16
    • Yohei SAKURABANobuyuki KiharaTakayoshi Kawaguchi
    • Yohei SAKURABANobuyuki KiharaTakayoshi Kawaguchi
    • H04B3/20
    • H04M9/082
    • 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.
    • 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. 发明申请
    • AUDIO PROCESSOR
    • 音频处理器
    • US20070206817A1
    • 2007-09-06
    • US11681025
    • 2007-03-01
    • Yohei SakurabaYasuhiko KatoNobuyuki Kihara
    • Yohei SakurabaYasuhiko KatoNobuyuki Kihara
    • H04B3/20H04M9/08
    • H04M9/082
    • 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.
    • 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
    • IMAGE PROCESSING APPARATUS, IMAGE PROCESSING METHOD, AND PROGRAM
    • 图像处理设备,图像处理方法和程序
    • US20140178049A1
    • 2014-06-26
    • US13823177
    • 2012-08-01
    • Nobuyuki KiharaYohei SakurabaTakeshi YamaguchiYasuhiko Kato
    • Nobuyuki KiharaYohei SakurabaTakeshi YamaguchiYasuhiko Kato
    • G11B27/036
    • H04N5/262G03B31/00G10L15/26G10L2015/088H04N5/91H04N9/74H04N9/8211
    • 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.
    • 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。
    • 7. 发明授权
    • Audio processor
    • 音频处理器
    • US08107617B2
    • 2012-01-31
    • US11681025
    • 2007-03-01
    • Yohei SakurabaYasuhiko KatoNobuyuki Kihara
    • Yohei SakurabaYasuhiko KatoNobuyuki Kihara
    • H04M9/08
    • H04M9/082
    • 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.
    • 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。
    • 8. 发明申请
    • SOUND SIGNAL PROCESSOR AND DELAY TIME SETTING METHOD
    • 声信号处理器和延迟时间设置方法
    • US20100183163A1
    • 2010-07-22
    • US12663332
    • 2008-06-05
    • Takeshi MatsuiYasuhiko KatoNobuyuki KiharaHideki KishiYasuhiro KodamaYohei Sakuraba
    • Takeshi MatsuiYasuhiko KatoNobuyuki KiharaHideki KishiYasuhiro KodamaYohei Sakuraba
    • H04B3/20
    • H04R3/02H04M9/082
    • 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.
    • 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。
    • 10. 发明授权
    • Audio reproducing system and method thereof
    • 音频再现系统及其方法
    • US08494192B2
    • 2013-07-23
    • US11699810
    • 2007-01-30
    • Teppei YokotaNobuyuki Kihara
    • Teppei YokotaNobuyuki Kihara
    • H04R5/02
    • H04R3/12H04R1/028H04S3/008H04S2420/01
    • An audio reproducing system includes a pair of speaker units, a mounting unit for mounting the pair of speaker units, without being attached to a baffle board, to the vicinity of a listener's ears in a manner such that sounds emitted from the front and back of a diaphragm of each speaker unit are mixed, and an audio signal output unit for virtual sound imaging an input audio signal and outputting the virtual sound imaged signal to the pair of speaker units in a manner such that the listener listens to a sound reproduced by the pair of speaker units feeling as if the sound is emitted from a different speaker device.
    • 音频再现系统包括一对扬声器单元,用于将一对扬声器单元安装到挡板上的安装单元,以使得从前后方向发出的声音到达听筒耳朵附近 混合每个扬声器单元的振膜,以及音频信号输出单元,用于虚拟声音成像输入音频信号,并以这样的方式将虚拟声音成像信号输出到一对扬声器单元,使得听者听到由 一对扬声器单元感觉好像从不同的扬声器装置发出声音。