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1. 发明授权

US08311237B2 Howling suppression apparatus and computer readable recording medium 失效
标题翻译：啸声抑制装置和计算机可读记录介质
公开(公告)号：US08311237B2
公开(公告)日：2012-11-13
申请号：US12360974
申请日：2009-01-28
申请人： Hirobumi Tanaka , Hiraku Okumura
发明人： Hirobumi Tanaka , Hiraku Okumura
IPC分类号： H04B15/00
CPC分类号： H04R3/02
摘要： A howling suppression apparatus suppresses a howling caused in an acoustic system including a sound collection device and a sound emission device. An estimation part generates an estimated signal by estimating a feedback sound reaching the sound collection device from the sound emission device. An adjustment part generates an estimated signal by adjusting the estimated signal. A spectrum subtraction part generates an acoustic signal using a result of subtracting a frequency spectrum of the estimated signal from a frequency spectrum of an acoustic signal. A filter part generates an acoustic signal by suppressing a component of a frequency band including a howling frequency F among the acoustic signal. An acoustic signal in which the acoustic signal is amplified by an amplifier is supplied to the sound emission device.
摘要翻译：啸声抑制装置抑制在包括声音收集装置和声音发射装置的声学系统中引起的啸声。估计部通过从声音发射装置估计到达声音采集装置的反馈声音来生成估计信号。调整部通过调整估计信号来生成估计信号。频谱减法部分使用从声信号的频谱中减去估计信号的频谱的结果来产生声信号。滤波器部分通过抑制声信号中包括啸叫频率F的频带的分量来产生声信号。将声音信号由放大器放大的声音信号提供给声音发射装置。

2. 发明授权

US07697696B2 Audio amplification apparatus with howling canceler 失效
标题翻译：带啸叫消音器的音频放大装置
公开(公告)号：US07697696B2
公开(公告)日：2010-04-13
申请号：US11326166
申请日：2006-01-04
申请人： Hiraku Okumura
发明人： Hiraku Okumura
IPC分类号： H04B3/20
CPC分类号： H04R3/02
摘要： In an audio amplification apparatus connected to a microphone and a speaker, an audio amplification circuit amplifies an audio signal and inputs the amplified audio signal to the speaker. A howling canceller has an adaptive filter which is set with a filter coefficient based on the audio signal input to the speaker and a residual signal so as to simulate a feedback transmission path from the speaker to the microphone such that the adaptive filter processes the audio signal to produce a simulation signal. The residual signal is obtained by subtracting the simulation signal from an input audio signal inputted from the microphone and fed to the audio amplification circuit. An internal sound source generates the audio signal and inputs the audio signal to the speaker. A sound source determination portion determines whether or not the input audio signal contains an external audio signal provided from an external sound source other than the audio signal fed back from the speaker to the microphone, and controls update of the filter coefficient in accordance with a result of the determination.
摘要翻译：在连接到麦克风和扬声器的音频放大装置中，音频放大电路放大音频信号并将放大的音频信号输入到扬声器。啸声消除器具有基于输入到扬声器的音频信号的滤波器系数和残余信号设置的自适应滤波器，以便模拟从扬声器到麦克风的反馈传输路径，使得自适应滤波器处理音频信号以产生模拟信号。残留信号是通过从麦克风输入的输入音频信号中减去模拟信号而得到的，并馈送给音频放大电路。内部声源产生音频信号并将音频信号输入到扬声器。音源确定部分确定输入音频信号是否包含从扬声器向麦克风反馈的音频信号以外的外部声源提供的外部音频信号，并根据结果控制滤波器系数的更新的决心。

3. 发明申请

US20060172272A1 Audio amplification apparatus with howling canceler 失效
标题翻译：带啸叫消音器的音频放大装置
公开(公告)号：US20060172272A1
公开(公告)日：2006-08-03
申请号：US11326166
申请日：2006-01-04
申请人： Hiraku Okumura
发明人： Hiraku Okumura
IPC分类号： G09B5/00
CPC分类号： H04R3/02
摘要： In an audio amplification apparatus connected to a microphone and a speaker, an audio amplification circuit amplifies an audio signal and inputs the amplified audio signal to the speaker. A howling canceler has an adaptive filter which is set with a filter coefficient based on the audio signal input to the speaker and a residual signal so as to simulate a feedback transmission path from the speaker to the microphone such that the adaptive filter processes the audio signal to produce a simulation signal. The residual signal is obtained by subtracting the simulation signal from an input audio signal inputted from the microphone and fed to the audio amplification circuit. An internal sound source generates the audio signal and inputs the audio signal to the speaker. A sound source determination portion determines whether or not the input audio signal contains an external audio signal provided from an external sound source other than the audio signal fed back from the speaker to the microphone, and controls update of the filter coefficient in accordance with a result of the determination.
摘要翻译：在连接到麦克风和扬声器的音频放大装置中，音频放大电路放大音频信号并将放大的音频信号输入到扬声器。啸叫消除器具有自适应滤波器，其基于输入到扬声器的音频信号设置滤波器系数和残余信号，以便模拟从扬声器到麦克风的反馈传输路径，使得自适应滤波器处理音频信号以产生模拟信号。残留信号是通过从麦克风输入的输入音频信号中减去模拟信号而得到的，并馈送给音频放大电路。内部声源产生音频信号并将音频信号输入到扬声器。音源确定部分确定输入音频信号是否包含从扬声器向麦克风反馈的音频信号以外的外部声源提供的外部音频信号，并根据结果控制滤波器系数的更新的决心。

4. 发明申请

US20060008076A1 Adaptive hauling canceller 有权
标题翻译：自适应拖运消除器
公开(公告)号：US20060008076A1
公开(公告)日：2006-01-12
申请号：US11175936
申请日：2005-07-06
申请人： Hiraku Okumura , Hiroaki Fujita
发明人： Hiraku Okumura , Hiroaki Fujita
IPC分类号： H04M9/08
CPC分类号： H04R3/02
摘要： An adaptive hauling canceller has a plurality of adaptive filters. A delay adds a time delay of an acoustic feedback path to an electric signal fed from an amplifier of a sound-reinforcement system. Each adaptive filter filters the output signal of the delay with a filter coefficient, which is periodically updated at an update interval. The update interval of each adaptive filter is set to decrease successively from a first one to a last one of the adaptive filters. Adders are arranged in correspondence to the adaptive filters in series between a microphone and the amplifier. Each adder subtracts the output signal of the corresponding adaptive filter from an output signal fed from a preceding adder to thereby provide an output signal to a succeeding adder. The output signal from each adder is inputted into the corresponding adaptive filter. The audio signal from the microphone is inputted to the first adder, while the output signal from the last adder is inputted through the amplifier to the speaker and to the delay as the electric signal. The filter coefficient of each adaptive filter is updated so as to simulate a transfer function of the acoustic feedback path based on the output signals of the corresponding adder and the delay.
摘要翻译：自适应牵引消除器具有多个自适应滤波器。延迟将声反馈路径的时间延迟添加到从扩声系统的放大器馈送的电信号。每个自适应滤波器以延迟的滤波器系数对延迟的输出信号进行滤波，该滤波器系数以更新间隔周期性地更新。每个自适应滤波器的更新间隔被设置为从第一个到最后一个自适应滤波器连续地减小。加法器与麦克风和放大器之间串联的自适应滤波器对应地布置。每个加法器从前一加法器输出的输出信号中减去对应的自适应滤波器的输出信号，从而向后一加法器提供输出信号。来自每个加法器的输出信号被输入到相应的自适应滤波器中。来自麦克风的音频信号被输入到第一加法器，而来自最后一个加法器的输出信号通过放大器输入到扬声器和作为电信号的延迟。每个自适应滤波器的滤波器系数被更新，以便基于对应加法器的输出信号和延迟来模拟声反馈路径的传递函数。

5. 发明授权

US08767951B2 Talk deciding system 有权
标题翻译：谈判决定系统
公开(公告)号：US08767951B2
公开(公告)日：2014-07-01
申请号：US11895231
申请日：2007-08-22
申请人： Satoshi Sekine , Hiraku Okumura
发明人： Satoshi Sekine , Hiraku Okumura
IPC分类号： H04M9/08
CPC分类号： H04M9/082
摘要： A talk deciding system includes a receiving portion which receives far-end side decision information indicating a talking state on a far-end side, an inputting portion which inputs an emitted signal from the far-end side, a speaker which emits the emitted signal being input into the inputting portion, a microphone which picks up a voice on a near-end side and outputs a picked-up signal, and a deciding portion which preliminarily decides a talking state on the near-end side based on the picked-up signal being output from the microphone and the emitted signal being input into the speaker, and generates near-end side decision information indicating a talking state on the near-end side. The deciding portion decides the talking state on the near-end side based on the far-end side decision information being received at the receiving portion and the near-end side decision information.
摘要翻译：话音判定系统包括：接收部，其接收表示远端侧的通话状态的远端侧判定信息，输入从远端侧发出的信号的输入部;发出发射信号的扬声器输入到输入部分中，麦克风拾取近端侧的声音并输出拾取信号;以及判定部，其基于拾取信号预先判定近端侧的通话状态从麦克风输出，并且发射的信号被输入到扬声器中，并产生指示近端侧的通话状态的近端侧判定信息。决定部基于在接收部接收的远端判定信息和近端侧判定信息，决定近端侧的通话状态。

6. 发明申请

US20100166213A1 Anti-Feedback Device and Anti-Feedback Method 有权
标题翻译：反反馈设备和反馈方法
公开(公告)号：US20100166213A1
公开(公告)日：2010-07-01
申请号：US12645203
申请日：2009-12-22
申请人： Hiraku OKUMURA , Hirobumi Tanaka
发明人： Hiraku OKUMURA , Hirobumi Tanaka
IPC分类号： H04B15/00
CPC分类号： H04R3/02 , H04R2430/03
摘要： An anti-feedback device includes an anti-feedback filter provided in a closed loop, The anti-feedback device down-samples a signal of specific band selected from an output signal of an adaptive target signal transfer system and a signal of the same band selected from an input signal of the transfer system, and a filtering coefficient of the adaptive filter is updated by use of the down-sampled signals. The filter controller controls a filtering characteristic of the anti-feedback filter so that a peak gain of a frequency of an amplitude characteristic within a specific band of a closed loop determined from the filtering coefficient of the adaptive filter is suppressed. Moreover, the filter controller estimates a gain of the closed loop outside the specific band from the amplitude characteristic in the specific band and controls the amount of suppression of the anti-feedback filter, outside the band in accordance with a result of estimation.
摘要翻译：反反馈装置包括设置在闭环中的反反馈滤波器。反反馈装置对从自适应目标信号传送系统的输出信号选择的特定频带的信号和选择的相同频带的信号进行下采样并且通过使用下采样信号来更新自适应滤波器的滤波系数。滤波器控制器控制反反馈滤波器的滤波特性，使得根据自适应滤波器的滤波系数确定的闭环的特定频带内的振幅特性的频率的峰值增益被抑制。此外，滤波器控制器根据特定频带的幅度特性估计特定频带外的闭环的增益，并根据估计结果控制频带外的反反馈滤波器的抑制量。

7. 发明申请

US20080056483A1 Talk deciding system 有权
标题翻译：谈判决定系统
公开(公告)号：US20080056483A1
公开(公告)日：2008-03-06
申请号：US11895231
申请日：2007-08-22
申请人： Satoshi Sekine , Hiraku Okumura
发明人： Satoshi Sekine , Hiraku Okumura
IPC分类号： H04M9/08
CPC分类号： H04M9/082
摘要： A talk deciding system includes a receiving portion which receives far-end side decision information indicating a talking state on a far-end side, an inputting portion which inputs an emitted signal from the far-end side, a speaker which emits the emitted signal being input into the inputting portion, a microphone which picks up a voice on a near-end side and outputs a picked-up signal, and a deciding portion which preliminarily decides a talking state on the near-end side based on the picked-up signal being output from the microphone and the emitted signal being input into the speaker, and generates near-end side decision information indicating a talking state on the near-end side. The deciding portion decides the talking state on the near-end side based on the far-end side decision information being received at the receiving portion and the near-end side decision information.
摘要翻译：话音判定系统包括：接收部，其接收表示远端侧的通话状态的远端侧判定信息，输入从远端侧发出的信号的输入部;发出发射信号的扬声器输入到输入部分中，麦克风拾取近端侧的声音并输出拾取信号;以及判定部，其基于拾取信号预先判定近端侧的通话状态从麦克风输出，并且发射的信号被输入到扬声器中，并产生指示近端侧的通话状态的近端侧判定信息。决定部基于在接收部接收的远端判定信息和近端侧判定信息，决定近端侧的通话状态。

8. 发明申请

US20060153400A1 Microphone and sound amplification system 审中-公开
标题翻译：麦克风和声音放大系统
公开(公告)号：US20060153400A1
公开(公告)日：2006-07-13
申请号：US11330549
申请日：2006-01-11
申请人： Hiroaki Fujita , Hiraku Okumura
发明人： Hiroaki Fujita , Hiraku Okumura
IPC分类号： H04B3/20 , H04R3/00
CPC分类号： H04R3/02
摘要： Microphone includes: a microphone element; a simulative feedback signal generation section that generates a simulative feedback signal simulating a feedback signal generated by a sound, produced via a speaker, returning the microphone element; and an arithmetic operator that subtracts the simulative feedback signal, generated by the simulative feedback signal generation section, from a sound signal collected by the microphone element, to thereby output the subtraction result as a residual signal. The residual signal output by the arithmetic operator is supplied to an amplifier device of the speaker as an output signal of the microphone. The simulative feedback signal generation section includes a delay circuit that delays the residual signal, output by the arithmetic operator, by a given time, and an adaptive filter that generates the simulative feedback signal by filtering a previous residual signal delayed by the delay circuit. The adaptive filter updates a filter coefficient on the basis of the previous residual signal delayed by the delay circuit and a current residual signal output by the arithmetic operator.
摘要翻译：麦克风包括：麦克风元件; 模拟反馈信号产生部分，其产生模拟由扬声器产生的声音产生的反馈信号的模拟反馈信号，使麦克风元件返回; 以及算术运算器，从由所述麦克风元件收集的声音信号中减去由所述模拟反馈信号生成部生成的所述模拟反馈信号，从而输出所述减法结果作为残差信号。由算术运算器输出的残留信号作为麦克风的输出信号提供给扬声器的放大器装置。模拟反馈信号生成部分包括将由算术运算器输出的残差信号延迟给定时间的延迟电路和通过对延迟电路延迟的先前残差信号进行滤波来产生模拟反馈信号的自适应滤波器。自适应滤波器基于由延迟电路延迟的先前残差信号和由算术运算器输出的电流剩余信号来更新滤波器系数。

9. 发明授权

US08218788B2 Anti-feedback device and anti-feedback method 有权
标题翻译：反反馈装置和反馈反馈方法
公开(公告)号：US08218788B2
公开(公告)日：2012-07-10
申请号：US12645203
申请日：2009-12-22
申请人： Hiraku Okumura , Hirobumi Tanaka
发明人： Hiraku Okumura , Hirobumi Tanaka
IPC分类号： H04B15/00 , H04R27/00 , H04R25/00
CPC分类号： H04R3/02 , H04R2430/03
摘要： An anti-feedback device includes an anti-feedback filter provided in a closed loop. The anti-feedback device down-samples a signal of specific band selected from an output signal of an adaptive target signal transfer system and a signal of the same band selected from an input signal of the transfer system, and a filtering coefficient of the adaptive filter is updated by use of the down-sampled signals. The filter controller controls a filtering characteristic of the anti-feedback filter so that a peak gain of a frequency of an amplitude characteristic within a specific band of a closed loop determined from the filtering coefficient of the adaptive filter is suppressed. Moreover, the filter controller estimates a gain of the closed loop outside the specific band from the amplitude characteristic in the specific band and controls the amount of suppression of the anti-feedback filter outside the band in accordance with a result of estimation.
摘要翻译：反反馈装置包括设置在闭环中的反反馈滤波器。反馈装置对从自适应目标信号传送系统的输出信号选择的特定频带的信号和从传送系统的输入信号选择的相同频带的信号进行下采样，并且自适应滤波器的滤波系数通过使用下采样信号进行更新。滤波器控制器控制反反馈滤波器的滤波特性，使得根据自适应滤波器的滤波系数确定的闭环的特定频带内的振幅特性的频率的峰值增益被抑制。此外，滤波器控制器根据特定频带的幅度特性估计特定频带外的闭环的增益，并且根据估计结果控制频带外的反反馈滤波器的抑制量。

10. 发明授权

US07925008B2 Multi-channel echo cancel method, multi-channel sound transfer method, stereo echo canceller, stereo sound transfer apparatus and transfer function calculation apparatus 失效
公开(公告)号：US07925008B2
公开(公告)日：2011-04-12
申请号：US11825301
申请日：2007-07-05
申请人： Toru Hirai , Mikio Tohyama , Hiraku Okumura , Yoshikazu Honji , Hiroshi Iriyama , Koji Niimi
发明人： Toru Hirai , Mikio Tohyama , Hiraku Okumura , Yoshikazu Honji , Hiroshi Iriyama , Koji Niimi
IPC分类号： H04M9/08
CPC分类号： H04R3/02 , H04M9/082
摘要： Stereo sound signals are reproduced directly from loudspeakers (SP(L), SP(R)). By using a sum signal and a difference signal of the stereo sound signals as a reference signal, and according to a cross spectrum calculation of the reference signal with a microphone-collected sound signal, calculation is performed to obtain transfer functions of four sound transfer systems between the loudspeakers (SP(L), SP(R)) and microphones (MC(L), MC(R)). The transfer functions obtained are subjected to inverse Fourier transform to obtain impulse responses, which are set in filter means (40-1 to 40-4) to create echo cancel signals and perform echo canceling. This solves the problem of an indefinite coefficient in the echo cancel technique of a multi-channel sound signal.

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