公开(公告)号：US08364484B2

公开(公告)日：2013-01-29

申请号：US12423209

申请日：2009-04-14

Applicant: Takehide Yano ,  Tadashi Amada ,  Kazunori Imoto ,  Koichi Yamamoto

Inventor： Takehide Yano ,  Tadashi Amada ,  Kazunori Imoto ,  Koichi Yamamoto

IPC: G10L15/00 ,  G10L11/00 ,  G10L19/02 ,  H04M11/00 ,  H04M1/64

CPC classification number: G10L15/22 ,  G06F3/16 ,  G10L25/78

Abstract: An input voice detect is detected after starting a voice input waiting state; the detected voice is recognized; an elapsed time from the start of the voice input waiting state is counted; an informative sound which urges a user to input the voice is outputted when the elapsed time reaches a preset output set time; and the output of the informative sound is stopped when the elapsed time at the time of inputting the voice is shorter than the output set timedetect.

Abstract translation: 在开始语音输入等待状态之后检测输入语音检测; 检测到的声音被识别; 从语音输入等待状态开始的经过时间被计数; 当经过时间到达预设的输出设定时间时，输出促使用户输入声音的信息声音; 并且当输入语音时的经过时间比输出设定时间短时，信息声音的输出停止。

公开(公告)号：US07706550B2

公开(公告)日：2010-04-27

申请号：US11028317

申请日：2005-01-04

Applicant: Tadashi Amada ,  Akinori Kawamura ,  Ryosuke Koshiba

Inventor： Tadashi Amada ,  Akinori Kawamura ,  Ryosuke Koshiba

IPC: H04B15/00

CPC classification number: G10L21/0208

Abstract: A noise estimation unit estimates a noise signal in an input signal. A section decision unit distinguishes a target signal section from a noise signal section in the input signal. A noise suppression unit suppresses the noise signal based on a first suppression coefficient from the input signal. A noise excess suppression unit suppresses the noise signal based on a second suppression coefficient from the input signal. The second suppression coefficient is larger than the first suppression coefficient. A switching unit switches between an output signal from the noise suppression unit and an output signal from the noise excess suppression unit based on a decision result of the section decision unit.

Abstract translation: 噪声估计单元估计输入信号中的噪声信号。 区段决定单元将输入信号中的目标信号区间与噪声信号区间区分开。 噪声抑制单元根据来自输入信号的第一抑制系数来抑制噪声信号。 噪声过剩抑制单元根据来自输入信号的第二抑制系数来抑制噪声信号。 第二抑制系数大于第一抑制系数。 开关单元基于区段判定单元的判定结果，在来自噪声抑制单元的输出信号和来自噪声过剩抑制单元的输出信号之间切换。

公开(公告)号：US20090326944A1

公开(公告)日：2009-12-31

申请号：US12423209

申请日：2009-04-14

Applicant: Takehide Yano ,  Tadashi Amada ,  Kazunori Imoto ,  Koichi Yamamoto

Inventor： Takehide Yano ,  Tadashi Amada ,  Kazunori Imoto ,  Koichi Yamamoto

IPC: G10L15/00

CPC classification number: G10L15/22 ,  G06F3/16 ,  G10L25/78

Abstract: An input voice detect is detected after starting a voice input waiting state; the detected voice is recognized; an elapsed time from the start of the voice input waiting state is counted; an informative sound which urges a user to input the voice is outputted when the elapsed time reaches a preset output set time; and the output of the informative sound is stopped when the elapsed time at the time of inputting the voice is shorter than the output set timedetect.

Abstract translation: 在开始语音输入等待状态之后检测输入语音检测; 检测到的声音被识别; 从语音输入等待状态开始的经过时间被计数; 当经过时间到达预设的输出设定时间时，输出促使用户输入声音的信息声音; 并且当输入语音时的经过时间比输出设定时间短时，信息声音的输出停止。

公开(公告)号：US20090048824A1

公开(公告)日：2009-02-19

申请号：US12192670

申请日：2008-08-15

Applicant: Tadashi Amada

Inventor： Tadashi Amada

IPC: G06F17/21

CPC classification number: G10L21/0208 ,  G10L25/27 ,  G10L2021/02166 ,  H04R3/005

Abstract: An audible signal process method includes preparing, in at least one dictionary, a plurality of weighting factors each learned to optimize evaluation function established by a weighted learning audible signal and a target speech signal corresponding to the learning audible signal and used for weighting, estimating a noise component included in the input audible signal, calculating a feature quantity depending upon the noise component of the input audible signal, selecting a weighting factor corresponding to the feature quantity from the dictionary, and weighting the input audible signal using the selected weighting factor to generate a processed output audible signal.

Abstract translation: 声音信号处理方法包括在至少一个字典中准备多个加权因子，每个加权因子被学习以优化由加权学习可听信号和对应于学习可听信号的目标语音信号建立的评估函数，并用于加权，估计 包括在输入声音信号中的噪声分量，根据输入的可听信号的噪声分量来计算特征量，从词典中选择与特征量对应的加权因子，以及使用所选择的加权因子对输入的可听信号进行加权以产生 经处理的输出可听信号。

公开(公告)号：US20090043566A1

公开(公告)日：2009-02-12

申请号：US12176668

申请日：2008-07-21

Applicant: Tadashi Amada

Inventor： Tadashi Amada

IPC: G10L19/02

CPC classification number: G10L17/20 ,  G10L15/20 ,  G10L25/27 ,  G10L2021/02165

Abstract: A speech processing apparatus includes a plurality of microphones which receive speech produced by a first sound source to obtain first speech signals for a plurality of channels having one-to-one correspondence with the plurality of microphones, a calculation unit configured to calculate a first characteristic amount indicative of an inter-channel correlation of the first speech signals, a storage unit configured to store in advance a second characteristic amount indicative of an inter-channel correlation of second speech signals for the plurality of channels obtained by receiving speech produced by a second sound source by the plurality of microphones, and a collation unit configured to collate the first characteristic amount with the second characteristic amount to determine whether the first sound source matches with the second sound source.

Abstract translation: 语音处理装置包括：多个麦克风，其接收由第一声源产生的语音，以获得与多个麦克风一一对应的多个频道的第一语音信号;计算单元，被配置为计算第一特征 指示第一语音信号的信道间相关的量;存储单元，被配置为预先存储指示通过接收由第二语音信号产生的语音获得的多个频道获得的多个频道的第二语音信号的信道间相关性的第二特征量 多个麦克风的声源，以及对照单元，被配置为将第一特征量与第二特征量进行比较，以确定第一声源是否与第二声源匹配。

公开(公告)号：US20080310646A1

公开(公告)日：2008-12-18

申请号：US12135300

申请日：2008-06-09

Applicant: Tadashi Amada

Inventor： Tadashi Amada

IPC: H04R3/02

CPC classification number: G10L21/0208 ,  G10L19/008

Abstract: An audio signal processing method for processing input audio signals of plural channels includes calculating at least one feature quantity representing a difference between channels of input audio signals, selecting at least one weighting factor according to the feature quantity from at least one weighting factor dictionary prepared by learning beforehand, and subjecting the input audio signals of plural channels to signal processing including noise suppression and weighting addition using the selected weighting factor to generate output an output audio signal.

Abstract translation: 一种用于处理多个频道的输入音频信号的音频信号处理方法包括：计算表示输入音频信号的频道之间的差异的至少一个特征量，根据来自至少一个加权因子字典的特征量选择至少一个加权因子，所述加权因子由 并且使用所选择的加权因子对多个通道的输入音频信号进行信号处理，包括噪声抑制和加权相加，以产生输出音频信号。

公开(公告)号：US5819213A

公开(公告)日：1998-10-06

申请号：US791741

申请日：1997-01-30

Applicant: Masahiro Oshikiri ,  Tadashi Amada ,  Masami Akamine ,  Kimio Miseki

Inventor： Masahiro Oshikiri ,  Tadashi Amada ,  Masami Akamine ,  Kimio Miseki

IPC: G10L19/08 ,  G10L9/14

CPC classification number: G10L19/08

Abstract: A speech encoding method and apparatus including analyzing, using a codebook expressing speech parameters within a predetermined search range, an input speech signal in an audibility weighting filter corresponding to a pitch period longer than the search range of the codebook, and searching, from the codebook, on the basis of the analysis result, a combination of speech parameters by which the distortion of the input speech signal is minimized, and encoding the combination. The apparatus uses an adaptive codebook of pitch and a noise codebook. The codebooks search a group formed by extracting vectors of predetermined length from one original code vector, while sequentially shifting position so that the vectors overlap each other. The search group is further restricted and another preselection is made before the final search. Search is based on inversely convoluted, orthogonally transformed vectors.

Abstract translation: 一种语音编码方法和装置，包括使用在预定搜索范围内表达语音参数的码本来分析与音码周期相对应的音频周期的声音加权滤波器中的输入语音信号，并且从码本中搜索 基于分析结果，将输入语音信号的失真最小化的语音参数的组合以及编码该组合。 该装置使用音调和噪声码本的自适应码本。 码本搜索通过从一个原始码矢量提取预定长度的矢量而形成的组，同时依次移位位置使得矢量彼此重叠。 搜索组进一步限制，并在最终搜索之前进行另一预选。 搜索基于逆卷积正交变换载体。

公开(公告)号：US08630850B2

公开(公告)日：2014-01-14

申请号：US13240353

申请日：2011-09-22

Applicant: Kaoru Suzuki ,  Tadashi Amada

Inventor： Kaoru Suzuki ,  Tadashi Amada

IPC: G10L21/00

CPC classification number: H04B3/23 ,  H04M9/082

Abstract: In one embodiment, a signal processing method is disclosed. The method can perform filter processing of convoluting a tap coefficient in a first signal sequence to generate a second signal sequence. The method can subtract the second signal sequence from a third signal sequence to generate a fourth signal sequence. The third signal sequence includes an echo signal of the first signal sequence. The method can correct the tap coefficient in accordance with an amount of correction determined using a function. The function includes at least one of a first region and a second region, and has values limited. The first region is included in a negative value region of the fourth signal sequence. The second region is included in a positive value region of the fourth signal sequence.

Abstract translation: 在一个实施例中，公开了一种信号处理方法。 该方法可以执行在第一信号序列中卷积抽头系数的滤波处理以产生第二信号序列。 该方法可以从第三信号序列中减去第二信号序列以产生第四信号序列。 第三信号序列包括第一信号序列的回波信号。 该方法可以根据使用功能确定的校正量来校正抽头系数。 该功能包括第一区域和第二区域中的至少一个，并且具有限制的值。 第一区域被包括在第四信号序列的负值区域中。 第二区域被包括在第四信号序列的正值区域中。

公开(公告)号：US20120124603A1

公开(公告)日：2012-05-17

申请号：US13172643

申请日：2011-06-29

Applicant: Tadashi Amada

Inventor： Tadashi Amada

IPC: H04H60/33 ,  H04R29/00

CPC classification number: H04M3/567 ,  H04L12/1827 ,  H04M3/564 ,  H04M3/568 ,  H04M3/569 ,  H04M2201/41 ,  H04N7/142

Abstract: In one embodiment, there is provided an audio signal processor. The processor includes: a person position detector configured to detect each position of one or more persons present in a specific space; a grouping module configured to allocate the detected persons to one or more groups, wherein the number of the groups is less than a given number; a plurality of directionality controllers configured to control directionality of a microphone array; and a directionality setting module configured to set directionality of each of the groups in a corresponding one of the directionality controllers.

Abstract translation: 在一个实施例中，提供了一种音频信号处理器。 处理器包括：人员位置检测器，被配置为检测存在于特定空间中的一个或多个人的每个位置; 分组模块，被配置为将检测到的人员分配给一个或多个组，其中所述组的数量小于给定数量; 配置成控制麦克风阵列的方向性的多个方向性控制器; 以及方向性设定模块，其被配置为设置所述方向性控制器中的相应一个中的每个组的方向性。

公开(公告)号：US20110313763A1

公开(公告)日：2011-12-22

申请号：US13219844

申请日：2011-08-29

Applicant: Tadashi Amada

Inventor： Tadashi Amada

IPC: G10L15/20 ,  H04R3/00

CPC classification number: H04R3/005 ,  G10L25/78 ,  G10L2021/02165 ,  H04R2430/20

Abstract: According to one embodiment, a pickup signal processing apparatus includes microphones, a sound determining unit, a signal level calculating unit, a setting unit, and a calculating unit. The sound determining unit determines whether pickup signals picked up by the microphones are signals from a neighboring sound source or a background noise signal. The signal level calculating unit calculates the signal levels for the microphones. The setting unit sets a gain value of at least one microphone and reduces a difference between the signal levels for the microphones on the basis of the signal levels for the microphones, when determined that the pickup signal is the background noise signal. The calculating unit multiplies the pickup signal of the at least one microphone by the gain value set by the setting unit.

Abstract translation: 根据一个实施例，拾取信号处理装置包括麦克风，声音确定单元，信号电平计算单元，设置单元和计算单元。 声音确定单元确定由麦克风拾取的拾取信号是否是来自相邻声源的信号或背景噪声信号。 信号电平计算单元计算麦克风的信号电平。 当确定拾取信号是背景噪声信号时，设置单元设置至少一个麦克风的增益值，并且基于麦克风的信号电平来减小麦克风的信号电平之间的差异。 计算单元将至少一个麦克风的拾取信号乘以由设置单元设置的增益值。