公开(公告)号：US5602926A

公开(公告)日：1997-02-11

申请号：US115872

申请日：1993-09-03

申请人： Tadashi Ohashi ,  Kensaku Fujii ,  Juro Ohga

发明人： Tadashi Ohashi ,  Kensaku Fujii ,  Juro Ohga

IPC分类号： F01N1/00 ,  F01N1/06 ,  G10K11/16 ,  G10K11/178 ,  H03H17/00 ,  H03H17/02 ,  H03H21/00 ,  H04B3/04

CPC分类号： G10K11/1784 ,  G10K2210/3018 ,  G10K2210/30232 ,  G10K2210/3045 ,  G10K2210/3049 ,  G10K2210/506

摘要： A method and apparatus of determining the transfer characteristic in an active-noise-control system, which involves generating white noise at an end of a one-dimensional sound field that is defined by a linear ventilating system in which sound travels essentially parallel to the extended direction of the system; equalizing the transfer characteristic of the one-dimensional sound field and generating cancelling sound, according to an inverse of the transfer characteristic, to cancel the white noise and prevent noise being output from the other end of the one-dimensional sound field; continuously preventing the noise output and measuring the characteristic data of the one-dimensional sound field at, at least, one measuring point in the one-dimensional sound field; and calculating the transfer function of the one-dimensional sound field in the noise-output-prevented state, according to the characteristic data of the sound field.

摘要翻译： 一种确定有源噪声控制系统中的传输特性的方法和装置，其涉及在由线性通风系统限定的一维声场的末端产生白噪声，其中声音基本上平行于延伸的 系统方向; 根据传输特性的相反均衡一维声场的传递特性并产生消除声音，以消除白噪声并防止噪声从一维声场的另一端输出; 连续地防止噪声输出并测量一维声场中至少一个测量点的一维声场的特征数据; 以及根据声场的特征数据计算噪声输出防止状态下的一维声场的传递函数。

公开(公告)号：US06683960B1

公开(公告)日：2004-01-27

申请号：US09641660

申请日：2000-08-18

申请人： Kensaku Fujii ,  Juro Ohga

发明人： Kensaku Fujii ,  Juro Ohga

IPC分类号： A61F1106

CPC分类号： H03H21/0012 ,  G10K11/178 ,  G10K2210/112 ,  G10K2210/30232 ,  H03H2021/0089 ,  H03H2021/0092

摘要： An active noise control apparatus for transmitting, from a loud speaker, a secondary noise synthesized so as to have the same amplitude as and the opposite phase to a primary noise and for canceling the noise by acoustically overlapping the secondary noise. An overall system filter for simulating a characteristic of an overall system leading to an error detecting microphone from a noise detecting microphone is provided for the first and the second overall system filter. A second and a first noise control filter are connected to the first and the second overall system filter in cascade to form a noise control filter. A coefficient of an estimating noise transfer system filter, obtained when the difference between the differential output of both obtained when a white noise is applied to the circuit of cascade connections and the response difference of the first and the second overall system filter from a differential overall system filter becomes a minimum, is made the coefficient of the noise control filter.

摘要翻译： 一种有源噪声控制装置，用于从扬声器发送合成的第二噪声，以便与主要噪声具有相同的振幅和相位相位，并通过声学重叠二次噪声来消除噪声。 为第一和第二整个系统滤波器提供了用于模拟导致来自噪声检测麦克风的误差检测麦克风的整个系统的特性的整体系统滤波器。 第二和第一噪声控制滤波器级联连接到第一和第二整个系统滤波器，以形成噪声控制滤波器。 当将白噪声施加到级联连接的电路时获得的差分输出与第一和第二整个系统滤波器的响应差与差分整体的差异输出时获得的估计噪声传递系统滤波器的系数 系统滤波器成为最小值，是噪声控制滤波器的系数。

公开(公告)号：US5146470A

公开(公告)日：1992-09-08

申请号：US590376

申请日：1990-09-28

申请人： Kensaku Fujii ,  Juro Ohga ,  Hiroyuki Masuda

发明人： Kensaku Fujii ,  Juro Ohga ,  Hiroyuki Masuda

IPC分类号： H04B3/23

CPC分类号： H04B3/23 ,  H04B3/231 ,  H04B3/234

摘要： An adaptive digital filter for estimating a response characteristic of a signal path by monitoring a sampled output signal of the signal path, estimating a predetermined number of filter coefficients which represent the response characteristic of the signal path, and generating an estimated output signal of the signal path using a plurality of successive sampled input signals of the signal path and the estimated filter coefficients, where the estimation is carried out so that a difference between the output signal and the estimated output signal is reduced. Each of the filter coefficients is extracted by a low-pass filter where the low-pass filter coefficient in the low-pass filter can be set to a constant. Normalization can be carried out in either the input side or the output side of the low-pass filter. Otherwise, the low-pass filter coefficient may be set to 1-K.multidot.X.sub.j (m).sup.2 /r, where r is set equal to a norm of the sampled input signals in the beginning, and is then set to an integrated power of the sample input signals.

公开(公告)号：US5790440A

公开(公告)日：1998-08-04

申请号：US567632

申请日：1995-12-05

申请人： Kensaku Fujii ,  Juro Ohga

发明人： Kensaku Fujii ,  Juro Ohga

IPC分类号： H03H17/00 ,  H03H17/06 ,  H03H21/00 ,  H04B3/06 ,  G06F17/10

CPC分类号： H03H21/0012

摘要： An apparatus for estimating filter coefficients operates in a system which includes a filter simulating characteristics of an unknown signal transmission system based on a signal provided to the unknown signal transmission system and a response signal produced from the unknown signal transmission system. The apparatus includes an accumulation calculating section for accumulating a product of the signal provided to the unknown signal transmission system and a difference between the response signal from the unknown signal transmission system and an output signal of the filter for a given time period. The apparatus further includes a square-sum calculating section for accumulating a square of the signal provided to the unknown signal transmission system for the given time period. The apparatus also includes an adjusting-amounts simulating section for simulating coefficient adjusting amounts of the filter based on a ratio of an output of the accumulation calculating section to an output of the square-sum calculating section. In the apparatus, coefficients of the filter are adjusted by using the coefficient adjusting amounts simulated in the adjusting-amounts simulating section.

摘要翻译： 用于估计滤波器系数的装置在包括基于提供给未知信号传输系统的信号和由未知信号传输系统产生的响应信号的模拟未知信号传输系统的特性的滤波器的系统中操作。 该装置包括累积计算部分，用于累积提供给未知信号传输系统的信号的乘积和来自未知信号传输系统的响应信号与给定时间段之间的滤波器的输出信号之间的差。 该装置还包括一个平方和计算部分，用于在给定时间段内积累提供给未知信号传输系统的信号的平方。 该装置还包括调整量模拟部分，用于基于累积计算部分的输出与平方和计算部分的输出的比率来模拟滤波器的系数调整量。 在该装置中，通过使用在调整量模拟部分中模拟的系数调节量来调整滤波器的系数。

公开(公告)号：US5384843A

公开(公告)日：1995-01-24

申请号：US122787

申请日：1993-09-15

申请人： Hiroyuki Masuda ,  Kazutoshi Hosokawa ,  Kensaku Fujii ,  Juro Ohga

发明人： Hiroyuki Masuda ,  Kazutoshi Hosokawa ,  Kensaku Fujii ,  Juro Ohga

IPC分类号： H04B3/23 ,  H04M1/60 ,  H04M9/08

CPC分类号： H04B3/23 ,  H04M9/082

摘要： An acoustic echo canceler and a side-tone echo canceler provided in a hands-free telephone suppress an acoustic echo and a side-tone echo respectively with few error by providing automatic gain controllers and/or limiters in the hands-free telephone so that the acoustic echo canceler and the side-tone echo canceler operate in linear, and a directional characteristic of a microphone used in the hands-free telephone for reducing the acoustic echo is controlled by a microphone direction controller so that the microphone operates as an omnidirectional microphone when a level of a received signal of the hands-free telephone is less than a designated level and as a bidirectional microphone when the level exceeds the designated level, further gains of an output of the microphone increases in a low frequency range.

摘要翻译： 在免提电话机中提供的声学回波消除器和侧音回波消除器通过在免提电话中提供自动增益控制器和/或限制器来分别抑制声回声和侧音回波，具有很少的误差，使得 声回波消除器和侧音回波消除器以线性方式工作，并且用于减少声回波的免提电话中使用的麦克风的方向特性由麦克风方向控制器控制，使得话筒作为全向麦克风操作，当 当电平超过指定电平时，免提电话的接收信号的电平小于指定电平，并且作为双向麦克风，麦克风的输出的进一步增益在低频范围内增加。

公开(公告)号：US4908855A

公开(公告)日：1990-03-13

申请号：US219607

申请日：1988-07-15

申请人： Juro Ohga ,  Kensaku Fujii ,  Hiroyuki Masuda ,  Yuka Sato

发明人： Juro Ohga ,  Kensaku Fujii ,  Hiroyuki Masuda ,  Yuka Sato

IPC分类号： H04M1/19 ,  H04M1/60 ,  H04M9/08

CPC分类号： H04M9/08 ,  H04M9/085

摘要： An electronic telephone terminal having a transmitter and a receiver, both having an approximately linear acoustic-to-electric transduction characteristics, and having a surrounding noise suppression function, and including a variable attenuator for controlling a gain of a transmission system; a noise detection device for detecting surrounding noise; and a control device for controlling the variable attenuator in such a manner that when a sound pressure level input to the transmitter exceeds a predetermined threshold value, the gain is fixedly set to a constant value, and when the input sound pressure level is equal to or below the predetermined threshold level, the gain is controlled in response to a change in the surrounding noise level detected by the noise detection device.

摘要翻译： 一种具有发射器（1）和接收器（2）的电子电话终端，其具有大致线性的声 - 电转换特性，并且具有周围的噪声抑制功能，并且包括用于控制增益的可变衰减器（12） 的传输系统; 用于检测周围噪声的噪声检测装置（3）; 以及用于以这样的方式控制可变衰减器（12）的控制装置（63），即当输入到发射器（1）的声压级超过预定阈值时，增益被固定地设定为恒定值，并且当 输入声压级等于或低于预定阈值电平，响应于由噪声检测装置（3）检测到的周围噪声电平的变化来控制增益。

公开(公告)号：US5940499A

公开(公告)日：1999-08-17

申请号：US778194

申请日：1997-01-02

申请人： Kensaku Fujii ,  Juro Ohga ,  Hiroyuki Masuda

发明人： Kensaku Fujii ,  Juro Ohga ,  Hiroyuki Masuda

IPC分类号： H04B3/20 ,  H04M1/60 ,  H04M9/08 ,  H04M9/10 ,  H04R3/02 ,  H04M1/00

CPC分类号： H04M1/6016 ,  H04M9/08 ,  H04M9/10

摘要： A voice switch in a hands-free communications system performs selective attenuation with respect to respective voice signals being transmitted and received in respective transmitting and receiving paths, the transmitted voice signal having been converted from an audible voice by a microphone connected to the transmitting path and the received voice signal having been converted by a loudspeaker to an audible voice output. A detector selectively detects a currently transmitted voice signal at a normal level, a currently received voice signal at a normal level, and drops in the respective levels thereof to nil levels and provides corresponding outputs to a controller. When one of the transmitted and received voice signals is of a normal level and the detector newly detects the other thereof at a normal level, the controller selectively attenuates that other, newly-detected normal level voice signal. Further, when a current voice signal of a normal level drops to a nil level and, within a selected time interval, resumes its normal level and also the other voice signal is newly detected at a normal level, the controller preferentially attenuates the newly detected voice signal such that the resumed normal level voice signal is preferentially processed.

摘要翻译： 免提通信系统中的语音切换对相应的发送和接收路径中发送和接收的各个语音信号执行选择性衰减，所发送的语音信号已被连接到发送路径的麦克风的可听话音转换， 所接收的语音信号已被扬声器转换成声音输出。 检测器选择性地检测正常电平处的当前传输的语音信号，正常电平的当前接收到的语音信号，并将其相应电平下降到零电平，并向控制器提供相应的输出。 当发送和接收的语音信号中的一个是正常电平，并且检测器以正常水平新检测到另一个时，控制器选择性地衰减另一个新检测的正常电平语音信号。 此外，当正常电平的当前语音信号下降到零电平，并且在所选择的时间间隔内恢复其正常电平，并且在正常电平下重新检测另一语音信号时，控制器优先衰减新检测到的声音 信号，使得恢复正常电平话音信号被优先处理。

公开(公告)号：US5638311A

公开(公告)日：1997-06-10

申请号：US538446

申请日：1995-10-03

申请人： Kensaku Fujii ,  Juro Ohga

发明人： Kensaku Fujii ,  Juro Ohga

IPC分类号： G10K11/178 ,  H03H21/00 ,  H04B1/10 ,  H04B3/23 ,  H04M1/60 ,  H04M9/08 ,  G06F17/10

CPC分类号： G10K11/1784 ,  H03H21/0012 ,  H04M9/082 ,  G10K2210/108 ,  G10K2210/3012 ,  G10K2210/30232 ,  G10K2210/3026 ,  G10K2210/3027 ,  G10K2210/3053 ,  G10K2210/503 ,  G10K2210/504 ,  G10K2210/505 ,  H03H2021/007 ,  H03H2021/0074 ,  H03H2021/0089

摘要： An estimation apparatus predicts filter coefficients for an adaptive filter, the response of which emulates the signal transmission characteristics of a known signal. The response thereto is sent to a signal transmission system of unknown characteristics, enabling execution of calculations without invalidating coefficient updating, even when there is a limit on the word length for processings. To achieve this, a sum of products calculation unit calculates the sum of products of the residual difference in response and the signal which is sent to the signal transmission system. A sum of squares calculating unit calculates the sum of the squares of the signal sent to the signal transmission system over a prescribed period of time. An updating amount calculation unit calculates the filter coefficient updating amounts from the ratio of the results from the sum of the products calculating unit to the results of the sum of the squares calculating unit. Filter coefficients are updated using the coefficient updating amounts calculated by the updating amount calculation unit.

摘要翻译： 估计装置预测自适应滤波器的滤波器系数，其自适应滤波器的响应模拟已知信号的信号传输特性。 其响应被发送到具有未知特性的信号传输系统，使得能够执行计算而不使系数更新无效，即使对于处理的字长限制也是如此。 为了实现这一点，产品计算单元的总和计算出响应中的残差差和发送到信号传输系统的信号的乘积之和。 平方和计算单元计算在规定时间内发送到信号传输系统的信号的平方和。 更新量计算单元根据从乘积计算单元的总和到平方计算单元的总和的结果的比率来计算滤波器系数更新量。 使用由更新量计算单元计算的系数更新量来更新滤波器系数。

公开(公告)号：US5479502A

公开(公告)日：1995-12-26

申请号：US243518

申请日：1994-05-16

申请人： Juro Ohga ,  Hiroyuki Masuda ,  Kensaku Fujii ,  Yoshihiro Sakai

发明人： Juro Ohga ,  Hiroyuki Masuda ,  Kensaku Fujii ,  Yoshihiro Sakai

IPC分类号： H04M9/08

CPC分类号： H04M9/082

摘要： A speaking apparatus having a handfree conversation function, provided with an echo canceler which can cancel the echo caused by direct acoustic coupling between a speaker and a microphone positioned in a system having a casing, a ground surface and a speech switching circuit. In the apparatus, the amount of insertion attenuation of the transmitted and received signal is set with an upper limit of the amount of attenuation sufficient for cancelling the echo caused by the indirect acoustic coupling determined by the location where the speaking apparatus is used. In addition, the echo canceler and the speech switching circuit share optimum functions to obtain an excellent speaking quality apparatus with an inexpensive and small sized processing circuit.

摘要翻译： 一种具有免提通话功能的讲话装置，其具有回波消除器，该回波消除器可消除位于具有壳体，地面和语音切换电路的系统中的扬声器和麦克风之间的直接声耦合引起的回波。 在该装置中，发送和接收信号的插入衰减量被设置为足以消除由使用说话装置的位置确定的间接声耦合引起的回波的衰减量的上限。 此外，回波消除器和语音切换电路共享最佳功能，以获得具有廉价且小型处理电路的优秀的说话质量设备。

公开(公告)号：US5237562A

公开(公告)日：1993-08-17

申请号：US667663

申请日：1991-03-11

申请人： Kensaku Fujii ,  Juro Ohga ,  Hiroyuki Masuda ,  Yoshihiro Sakai

发明人： Kensaku Fujii ,  Juro Ohga ,  Hiroyuki Masuda ,  Yoshihiro Sakai

IPC分类号： G01N15/00 ,  G01H3/00 ,  H04B3/23 ,  H04M9/08

CPC分类号： H04B3/234 ,  H04M9/082

摘要： A method for detecting a transition of an echo path used for an estimation of the transfer function of a system by using an adaptive filter, from an echo caused by a signal transmitted through an input terminal of the system at which the response of the transmitted signal is received. The method includes the steps of calculating the amount of the whole or the first delay portion of an impulse response of the system, calculating the amount of the remaining delay portion of the impulse response of the system, calculating the ratio between the amount of the whole or the first delay portion of the impulse response and the amount of the remaining delay portion of the impulse response, and deciding an occurrence of a transition of the characteristics of the system based on the calculated ratio. The transition of the characteristics of the system is discriminated from a hindering signal produced in the system in the decision making process.

摘要翻译： 一种用于通过使用自适应滤波器来检测用于估计系统的传递函数的回波路径的转换的方法，该回波路径是由通过系统的输入端发送的信号引起的， 被收到。 该方法包括以下步骤：计算系统的脉冲响应的整个或第一延迟部分的量，计算系统的脉冲响应的剩余延迟部分的量，计算整个数量之间的比率 或脉冲响应的第一延迟部分和脉冲响应的剩余延迟部分的量，并且基于所计算的比率来确定系统的特性的转变的发生。 系统特性的转变与决策过程中系统产生的阻碍信号有所区别。