公开(公告)号：US20090296526A1

公开(公告)日：2009-12-03

申请号：US12472469

申请日：2009-05-27

Applicant: Tadashi Amada

Inventor： Tadashi Amada

IPC: G01S15/00

CPC classification number: G01S3/8034 ,  G10L15/28 ,  G10L25/48 ,  G10L2021/02166

Abstract: An acoustic treatment apparatus obtains a first output signal by performing filtering for forming a directivity in a first direction for received sound signals of sound receivers, obtains a second output signal by performing filtering for forming a directivity in a second direction different from the first direction for received sound signals of sound receivers, obtains a strength ratio between a strength of the first output signal and a strength of the second output signal, and estimates a sound source direction on the basis of the strength ratio.

Abstract translation: 声学处理装置通过对声音接收机的接收声音信号进行第一方向的方向性的滤波，得到第一输出信号，通过对与第一方向不同的第二方向进行方向性的滤波，得到第二输出信号 接收到声音接收机的声音信号，获得第一输出信号的强度与第二输出信号的强度之间的强度比，并且基于强度比来估计声源方向。

公开(公告)号：US06611797B1

公开(公告)日：2003-08-26

申请号：US09488748

申请日：2000-01-21

Applicant: Tadashi Amada ,  Katsumi Tsuchiya

Inventor： Tadashi Amada ,  Katsumi Tsuchiya

IPC: G10L1910

CPC classification number: G10L19/12

Abstract: An input speech signal to an input terminal is supplied to a speech synthesizer section through a speech analyzer section and frequency parameter quantizer section to form a synthesis filter, and the input speech signal is expressed by quantized LPC coefficients representing the characteristics of the synthesis filter and an excitation signal for exciting the synthesis filter. In this case, in a pulse excitation section, a pulse position selector selects pulse position candidates from the integer pulse positions and non-integer pulse positions stored in a pulse position codebook, and an integer position pulse generator and non-integer position pulse generator respectively generate integer position pulses set at sampling points of the excitation signal and non-integer position pulses set at positions located between sampling points. These pulses are synthesized into a pulse train serving as a source of an excitation signal.

Abstract translation: 通过语音分析器部分和频率参数量化器部分向输入终端输入的语音信号提供给语音合成器部分以形成合成滤波器，并且输入语音信号由表示合成滤波器的特性的量化LPC系数表示， 用于激发合成滤波器的激励信号。 在这种情况下，在脉冲激励部中，脉冲位置选择器分别从存储在脉冲位置码本中的整数脉冲位置和非整数脉冲位置选择脉冲位置候补，分别选择整数位置脉冲发生器和非整数位置脉冲发生器 产生在激励信号的采样点设置的整数位置脉冲和设置在采样点之间的位置处的非整数位置脉冲。 这些脉冲被合成为用作激励信号的源的脉冲串。

公开(公告)号：US6167375A

公开(公告)日：2000-12-26

申请号：US39317

申请日：1998-03-16

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

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

IPC: G10L19/00 ,  G10L19/02

CPC classification number: G10L19/012 ,  G10L19/02

Abstract: A method for encoding speech wherein an input speech signal is separated by a component separator into a first component mainly constituted by speech and a second component mainly constituted by a background noise at each predetermined unit of time, a bit allocation selector selects bit allocation for each component based on the first and second components from among a plurality of predetermined candidates for bit allocation, a speech encoder and a noise encoder encode the first and second components from the component separator based on the bit allocation according to predetermined different methods for encoding, and a multiplexer multiplexes encoded data of the first and second components and information on the bit allocation and outputs them as transmitted encoded data.

Abstract translation: 一种编码语音的方法，其中输入语音信号由分量分离器分离成主要由语音构成的第一分量和主要由每个预定单位时间的背景噪声构成的第二分量，比特分配选择器选择每个 基于来自用于比特分配的多个预定候选中的第一和第二分量的分量，语音编码器和噪声编码器根据预定的不同编码方法基于比特分配对来自分量分离器的第一和第二分量进行编码，以及 复用器复用第一和第二分量的编码数据和关于比特分配的信息，并将其作为发送的编码数据输出。

公开(公告)号：US5926785A

公开(公告)日：1999-07-20

申请号：US911719

申请日：1997-08-15

Applicant: Masami Akamine ,  Tadashi Amada

Inventor： Masami Akamine ,  Tadashi Amada

IPC: G10L21/02 ,  G10L19/00 ,  G10L19/04 ,  G10L19/08 ,  G10L19/14 ,  H03M7/30 ,  G10L9/14

CPC classification number: G10L19/083 ,  G10L19/26

Abstract: A speech encoding method including generating a reconstruction speech vector by using a code vector extracted from a codebook storing a plurality of code vectors for encoding a speech signal. In addition an input speech signal to be encoded is used as a target vector to generate an error vector representing the error of the reconstruction speech vector with respect to the target vector, and the error vector is passed through a perceptual weighting filter having a transfer function including the inverse characteristics of the transfer function of a filter for emphasizing the spectrum of a reconstructed speech signal. Thus a weighted error vector is generated, the codebook for a code vector that minimizes the weighted error vector is searched, and an index corresponding to the code vector found as an encoding parameter is output.

Abstract translation: 一种语音编码方法，包括使用从存储用于编码语音信号的多个码矢量的码本提取的码矢量生成重构语音矢量。 此外，将要编码的输入语音信号用作目标矢量以产生表示相对于目标矢量的重构语音矢量的误差的误差矢量，并且将误差矢量通过具有传递函数的感知加权滤波器 包括用于强调重构语音信号的频谱的滤波器的传递函数的逆特性。 因此，产生加权误差向量，搜索最小化加权误差向量的码矢量的码本，并输出与作为编码参数发现的码矢量相对应的索引。

公开(公告)号：US08363821B2

公开(公告)日：2013-01-29

申请号：US12880772

申请日：2010-09-13

Applicant: Kaoru Suzuki ,  Koichi Yamamotop ,  Tadashi Amada

Inventor： Kaoru Suzuki ,  Koichi Yamamotop ,  Tadashi Amada

IPC: H04M9/08

CPC classification number: H04M9/082

Abstract: According to one embodiment, in response to a first acoustic signal output, a second acoustic signal is input. A filter unit is configured to generate a third acoustic signal by convoluting the first acoustic signal with coefficients. A subtraction unit is configured to generate a fourth acoustic signal by subtracting the third acoustic signal from the second acoustic signal. An estimation unit is configured to decide whether a sound volume of the first acoustic signal is below a predetermined threshold, and to set a sound volume of the second acoustic signal as a non-echo sound level when the sound volume of the first acoustic signal is below the predetermined threshold. A determination unit is configured to determine a step size to correct the coefficients using the non-echo sound level. A correction unit is configured to correct the coefficients using the step size.

Abstract translation: 根据一个实施例，响应于第一声信号输出，输入第二声信号。 滤波器单元被配置为通过用系数卷积第一声信号来产生第三声信号。 减法单元被配置为通过从第二声信号中减去第三声信号来产生第四声信号。 估计单元被配置为确定第一声信号的音量是否低于预定阈值，并且当第一声信号的音量是第一声信号时，将第二声信号的音量设置为非回波声级 低于预定阈值。 确定单元被配置为使用非回波声级来确定步长以校正系数。 校正单元被配置为使用步长校正系数。

公开(公告)号：US08229739B2

公开(公告)日：2012-07-24

申请号：US12176668

申请日：2008-07-21

Applicant: Tadashi Amada

Inventor： Tadashi Amada

IPC: G10L19/00

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：US08121310B2

公开(公告)日：2012-02-21

申请号：US12014473

申请日：2008-01-15

Applicant: Tadashi Amada

Inventor： Tadashi Amada

IPC: H04R3/00

CPC classification number: H04R3/005

Abstract: A plurality of sound receiving units is installed onto an equipment body. An initial information memory stores an initial direction of the equipment body in a terminal coordinate system based on the equipment body. An orientation detection unit detects an orientation of the equipment body in a world coordinate system based on a real space. A lock information output unit outputs lock information representing to rock the orientation. An orientation information memory stores the orientation detected when the lock information is output. A direction conversion unit converts the initial direction to a target sound direction in the world coordinate system by using the orientation stored in the orientation information memory. A directivity forming unit forms a directivity of the plurality of sound receiving units toward the target sound direction.

Abstract translation: 多个声音接收单元安装在设备主体上。 初始信息存储器基于设备主体将设备主体的初始方向存储在终端坐标系中。 姿态检测单元基于实际空间检测世界坐标系中的设备主体的取向。 锁定信息输出单元输出表示摇摆方向的锁定信息。 定位信息存储器存储当输出锁定信息时检测到的姿势。 方向转换单元通过使用存储在方向信息存储器中的方向将初始方向转换为世界坐标系中的目标声音方向。 指向性形成单元朝向目标声音方向形成多个声音接收单元的方向性。

公开(公告)号：US20070005350A1

公开(公告)日：2007-01-04

申请号：US11476024

申请日：2006-06-28

Applicant: Tadashi Amada

Inventor： Tadashi Amada

IPC: G10L21/00

CPC classification number: G10L25/27 ,  G10L2021/02082 ,  G10L2021/02166

Abstract: A sound signal processing method includes calculating a difference between every few ones of input multiple channel sound signals to obtain a plurality of characteristic quantities each indicating the difference, selecting a weighting factor from a weighting factor dictionary containing a plurality of weighting factors of a plurality of channels corresponding to the characteristic quantities, weighting the sound signals by using the selected weighting factor, and adding the weighted input sound signals to generate an output sound signal.

Abstract translation: 一种声音信号处理方法，包括：计算输入多声道声音信号的每一个之间的差分，得到各自表示差异的多个特征量，从包含多个声道信号的多个加权因子的加权系数字典中选择加权因子 对应于特征量的通道，通过使用所选择的加权因子对声音信号进行加权，以及将加权的输入声音信号相加以产生输出声音信号。

公开(公告)号：US5864798A

公开(公告)日：1999-01-26

申请号：US714260

申请日：1996-09-17

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

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

IPC: G10L19/14 ,  G10L9/00

CPC classification number: G10L19/26 ,  G10L25/12

Abstract: Adjusting the shape of a spectrum of a speech signal includes the steps of using a first filter with pole-zero transfer function A(z)/B(z) for subjecting a speech signal to a spectrum envelop emphasis and a second filter cascade-connected with the first filter, for compensating for a spectral tilt due to the first filter, independently deriving two filter coefficients used in the second filter for compensating for the spectral tilt from the pole-zero transfer function, and compensating for the spectral tilt corresponding to the pole-zero transfer function according to the derived filter coefficients.

Abstract translation: 调整语音信号的频谱的形状包括以下步骤：使用具有极零传递函数A（z）/ B（z）的第一滤波器来对语音信号进行频谱包络加强，并且第二滤波器级联连接 利用第一滤波器，用于补偿由于第一滤波器引起的频谱倾斜，独立地导出在第二滤波器中使用的两个滤波器系数，用于补偿来自极 - 零传递函数的频谱倾斜，以及补偿对应于 根据派生滤波系数的极零传递函数。

公开(公告)号：US5677986A

公开(公告)日：1997-10-14

申请号：US451174

申请日：1995-05-26

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

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

IPC: G10L19/08 ,  G06T9/00 ,  G10L19/00 ,  G10L19/12 ,  H03M7/30 ,  G10L7/02

CPC classification number: H03M7/3082 ,  G06T9/008

Abstract: A vector quantizing apparatus includes a first search section for obtaining an approximate vector X1 which is approximated to a desired vector R, a residual vector calculator for calculating a residual vector Rv from the desired vector R and the approximate vector X1, a weighting section for obtaining weighted vectors X2 to XN of code vectors x2 to xN, and a second search section for calculating an estimation value which is the magnitude of a projection vector of the residual vector Rv with respect to the vector space formed by the approximate vector X1 and the weighted vectors X2 to XN, and searching a code vector which maximizes this estimation value.

Abstract translation: 矢量量化装置包括用于获得近似于期望矢量R的近似矢量X1的第一搜索部分，用于从期望矢量R和近似矢量X1计算残差矢量Rv的残差矢量计算器，用于获得的加权部分 以及第二搜索部分，用于计算相对于由近似矢量X1形成的向量空间的残差矢量Rv的投影矢量的大小的估计值，以及加权矢量X 2的加权矢量 向量X2到XN，并且搜索最大化该估计值的码矢量。