公开(公告)号：US07340985B2

公开(公告)日：2008-03-11

申请号：US10997992

申请日：2004-11-29

Applicant: Ingvar Claesson ,  Thomas Lago ,  Lars Hakansson

Inventor： Ingvar Claesson ,  Thomas Lago ,  Lars Hakansson

IPC: B23Q17/12 ,  B23B27/00 ,  B23B29/00

CPC classification number: B23B27/002 ,  B23Q11/0032 ,  Y10S82/904 ,  Y10T82/10 ,  Y10T82/2502 ,  Y10T82/2533 ,  Y10T82/2585 ,  Y10T82/2595 ,  Y10T408/76 ,  Y10T409/304312

Abstract: A device and method for vibration control in a machine for cutting, said machine comprising a cutting tool supported by a tool holder. The device comprises a control unit and converting means which are connectible to the control unit and comprise a vibration sensor and an actuator. The actuator comprises an active element which converts an A.C. voltage supplied by the control unit to the actuator into dimensional changes. Said active element is adapted to be embedded in the body of the tool holder and in such manner that said dimensional changes impart bending to the body of the tool holder.

Abstract translation: 一种用于切割的机器中的振动控制的装置和方法，所述机器包括由工具架支撑的切割工具。 该装置包括可连接到控制单元并包括振动传感器和致动器的控制单元和转换装置。 致动器包括将由控制单元提供的交流电压转换为致动器的有源元件作为尺寸变化。 所述有源元件适于嵌入工具保持器的主体中并且以这种方式使得所述尺寸变化赋予工具保持器的主体弯曲。

公开(公告)号：US08693678B2

公开(公告)日：2014-04-08

申请号：US13384554

申请日：2010-06-17

Applicant: Fredric Lindström ,  Christian Schüldt ,  Ingvar Claesson

Inventor： Fredric Lindström ,  Christian Schüldt ,  Ingvar Claesson

IPC: H04M9/08

CPC classification number: H04M9/082 ,  G10L2021/02082

Abstract: The present invention relates to a device, such as a communication device, comprising an adaptive foreground filter configured to calculate a first echo estimation signal based on a first input signal, and an adaptive background filter being more rapidly adapting than the foreground filter and configured to calculate a second echo estimation signal based on said first input signal. Embodiments of the device further comprise damping control means for controlling damping of an echo-cancelled output signal. The device in various embodiments includes that the damping control means is configured to calculate a maximum echo estimation signal using both the first and the second echo estimation signals, and control the damping of the echo-cancelled output signal based on said maximum echo estimation signal and/or a signal derived from said maximum echo estimation signal.

Abstract translation: 本发明涉及一种诸如通信设备的设备，包括：自适应前景滤波器，被配置为基于第一输入信号计算第一回波估计信号，并且自适应背景滤波器比前台滤波器更快地适应并且被配置为 基于所述第一输入信号计算第二回波估计信号。 该装置的实施例还包括用于控制回波消除的输出信号的阻尼的阻尼控制装置。 各种实施例中的装置包括阻尼控制装置被配置为使用第一和第二回波估计信号来计算最大回波估计信号，并且基于所述最大回波估计信号来控制回波消除输出信号的衰减，以及 /或从所述最大回波估计信号得到的信号。

公开(公告)号：US06717991B1

公开(公告)日：2004-04-06

申请号：US09493265

申请日：2000-01-28

Applicant: Harald Gustafsson ,  Ulf Lindgren ,  Ingvar Claesson ,  Sven Nordholm

Inventor： Harald Gustafsson ,  Ulf Lindgren ,  Ingvar Claesson ,  Sven Nordholm

IPC: H04B1500

CPC classification number: H04R3/005

Abstract: Speech enhancement is provided in dual microphone noise reduction systems by including spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, when a far-mouth microphone is used in conjunction with a near-mouth microphone, it is possible to handle non-stationary background noise as long as the noise spectrum can continuously be estimated from a single block of input samples. The far-mouth microphone, in addition to picking up the background noise, also picks up the speaker's voice, albeit at a lower level than the near-mouth microphone. To enhance the noise estimate, a spectral subtraction stage is used to suppress the speech in the far-mouth microphone signal. To be able to enhance the noise estimate, a rough speech estimate is formed with another spectral subtraction stage from the near-mouth signal. Finally, a third spectral subtraction function is used to enhance the near-mouth signal by suppressing the background noise using the enhanced background noise estimate. A controller dynamically determines any or all of a first, second, and third subtraction factor for each of the first, second, and third spectral subtraction stages, respectively.

Abstract translation: 通过使用线性卷积，频谱减法增益函数的因果滤波和/或频谱依赖指数平均的频谱减法算法，在双麦克风降噪系统中提供语音增强。 根据示例性实施例，当远端麦克风与近嘴麦克风结合使用时，只要可以从单个输入样本块连续地估计噪声谱，就可以处理非平稳背景噪声。 远端麦克风除了吸收背景噪音外，还会吸收扬声器的声音，尽管距离近嘴麦克风较低。 为了增强噪声估计，使用频谱减法级来抑制远口麦克风信号中的语音。 为了能够增强噪声估计，粗略的语音估计是由近端信号的另一个谱减法阶段形成的。 最后，使用第三频谱减法函数，通过使用增强的背景噪声估计来抑制背景噪声来增强近嘴信号。 控制器分别动态地确定第一，第二和第三频谱减法级中的每一个的第一，第二和第三减法因子中的任何一个或全部。

公开(公告)号：US06549586B2

公开(公告)日：2003-04-15

申请号：US09289065

申请日：1999-04-12

Applicant: Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

Inventor： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

IPC: H04B1500

CPC classification number: H04R3/005

Abstract: Speech enhancement is provided in dual microphone noise reduction systems by including spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, when a far-mouth microphone is used in conjunction with a near-mouth microphone, it is possible to handle non-stationary background noise as long as the noise spectrum can continuously be estimated from a single block of input samples. The far-mouth microphone, in addition to picking up the background noise, also picks up the speaker's voice, albeit at a lower level than the near-mouth microphone. To enhance the noise estimate, a spectral subtraction stage is used to suppress the speech in the far-mouth microphone signal. To be able to enhance the noise estimate, a rough speech estimate is formed with another spectral subtraction stage from the near-mouth signal. Finally, a third spectral subtraction function is used to enhance the near-mouth signal by suppressing the background noise using the enhanced background noise estimate.

Abstract translation: 通过使用线性卷积，频谱减法增益函数的因果滤波和/或频谱依赖指数平均的频谱减法算法，在双麦克风降噪系统中提供语音增强。 根据示例性实施例，当远端麦克风与近嘴麦克风结合使用时，只要可以从单个输入样本块连续地估计噪声谱，就可以处理非平稳背景噪声。 远端麦克风除了吸收背景噪音外，还会吸收扬声器的声音，尽管距离近嘴麦克风较低。 为了增强噪声估计，使用频谱减法级来抑制远口麦克风信号中的语音。 为了能够增强噪声估计，粗略的语音估计是由近端信号的另一个谱减法阶段形成的。 最后，使用第三频谱减法函数，通过使用增强的背景噪声估计来抑制背景噪声来增强近嘴信号。

公开(公告)号：US06487257B1

公开(公告)日：2002-11-26

申请号：US09289554

申请日：1999-04-12

Applicant: Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

Inventor： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

IPC: H04B1500

CPC classification number: G10L21/0208

Abstract: For purposes of noise suppression, spectral subtraction filtering is performed in sample-wise fashion in the time domain using a time-domain representation of a spectral subtraction gain function computed in block-wise fashion in the frequency domain. By continuously performing time-domain filtering on a sample by sample basis, the disclosed methods and apparatus avoid block-processing delays associated with frequency-domain based spectral subtraction systems. Consequently, the disclosed methods and apparatus are particularly well suited for applications requiring very short processing delays. In applications where only stationary, low-energy background noise is present, computational complexity is reduced by generating a number of separate spectral subtraction gain functions during an initialization period, each gain function being suitable for one of several predefined classes of input signal (e.g., for one of several predetermined signal energy ranges), and thereafter fixing the several gain functions until the input signal characteristics change.

Abstract translation: 为了噪声抑制的目的，在时域中以采样方式执行频谱减法滤波，使用在频域中以块方式计算的频谱减法增益函数的时域表示。 通过以样本为基础连续执行时域滤波，所公开的方法和装置避免了与基于频域的谱减法系统相关的块处理延迟。 因此，所公开的方法和装置特别适用于需要非常短的处理延迟的应用。 在仅存在静止的低能量背景噪声的应用中，通过在初始化周期期间产生多个单独的频谱减法增益函数来减少计算复杂度，每个增益函数适用于若干预定义类别的输入信号之一（例如， 对于几个预定信号能量范围中的一个），然后固定多个增益函数直到输入信号特性改变。

公开(公告)号：US06459914B1

公开(公告)日：2002-10-01

申请号：US09084503

申请日：1998-05-27

Applicant: Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

Inventor： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

IPC: B13

CPC classification number: G10L21/0232 ,  G10L21/0264

Abstract: Methods and apparatus for providing speech enhancement in noise reduction systems include spectral subtraction algorithms using linear convolution, causal filtering and/or spectrum dependent exponential averaging of the spectral subtraction gain function. According to exemplary embodiments, successive blocks of a spectral subtraction gain function are averaged based on a discrepancy between an estimate of a spectral density of a noisy speech signal and an averaged estimate of a spectral density of a noise component of the noisy speech signal. The successive gain function blocks are averaged, for example, using controlled exponential averaging. Control is provided, for example, by making a memory of the exponential averaging inversely proportional to the discrepancy. Alternatively, the averaging memory can be made to increase in direct proportion with decreases in the discrepancy, while exponentially decaying with increases in the discrepancy to prevent audible voice shadows.

Abstract translation: 用于在降噪系统中提供语音增强的方法和装置包括使用线性卷积，频谱减法增益函数的因果滤波和/或频谱依赖指数平均的频谱减法算法。 根据示例性实施例，基于噪声语音信号的频谱密度的估计与噪声语音信号的噪声分量的频谱密度的平均估计之间的差异来平均频谱减法增益函数的连续块。 连续增益功能块被平均化，例如使用受控指数平均。 提供控制，例如，通过使指数平均值与差异成反比的记忆。 或者，可以使平均存储器随着差异的减小而成正比地增加，同时随着差异的增加呈指数衰减，以防止可听见的声音阴影。

公开(公告)号：US08351554B2

公开(公告)日：2013-01-08

申请号：US12301758

申请日：2006-06-05

Applicant: Nedelko Grbic ,  Ingvar Claesson ,  Per Eriksson

Inventor： Nedelko Grbic ,  Ingvar Claesson ,  Per Eriksson

IPC: H04B1/10

CPC classification number: H04R3/005 ,  G10L21/0208 ,  H04R25/407 ,  H04R2430/03

Abstract: The invention relates to an adaptive method of extracting at least of desired electro magnetic wave signals, sound wave signals (40, 42), and any other signals from a mixture of signals (40, 42, 44, 46) and suppressing noise and interfering signals to produce enhanced signals (50) corresponding to desired (10) signals, and an apparatus (70) therefore. It relies on the concept of at least one of an attenuation of input signals in each sub-band for signals in such a manner that all desired (10) signals are attenuated less than noise or interfering source signals, and/or an amplification of input signals in each sub-band for source signals in such a manner that all desired (10) signals are amplified, and that they are amplified more than the noise and interfering signals.

Abstract translation: 本发明涉及至少提取期望的电磁波信号，声波信号（40,42）和来自信号（40,42,44,46）的混合信号的任何其它信号并且抑制噪声和干扰的自适应方法 信号以产生对应于期望的（10）信号的增强信号（50），因此产生装置（70）。 它依赖于以下方式中的每个子带中的输入信号的衰减中的至少一个的概念：使得所有期望的（10）信号衰减小于噪声或干扰源信号，和/或输入的放大 每个子带中的信号以这样的方式使得所有期望的（10）信号被放大，并且它们被放大得比噪声和干扰信号多。

公开(公告)号：US20050109174A1

公开(公告)日：2005-05-26

申请号：US10997992

申请日：2004-11-29

Applicant: Ingvar Claesson ,  Thomas Lago ,  Lars Hakansson

Inventor： Ingvar Claesson ,  Thomas Lago ,  Lars Hakansson

IPC: B23Q15/12 ,  B23B1/00 ,  B23B27/00 ,  B23B29/12 ,  B23B37/00 ,  B23Q3/12 ,  B23Q5/22 ,  B23Q11/00 ,  B23B3/00

CPC classification number: B23B27/002 ,  B23Q11/0032 ,  Y10S82/904 ,  Y10T82/10 ,  Y10T82/2502 ,  Y10T82/2533 ,  Y10T82/2585 ,  Y10T82/2595 ,  Y10T408/76 ,  Y10T409/304312

Abstract: A device and method for vibration control in a machine for cutting, said machine comprising a cutting tool supported by a tool holder. The device comprises a control unit and converting means which are connectible to the control unit and comprise a vibration sensor and an actuator. The actuator comprises an active element which converts an A.C. voltage supplied by the control unit to the actuator into dimensional changes. Said active element is adapted to be embedded in the body of the tool holder and in such manner that said dimensional changes impart bending to the body of the tool holder.

Abstract translation: 一种用于切割的机器中的振动控制的装置和方法，所述机器包括由工具架支撑的切割工具。 该装置包括可连接到控制单元并包括振动传感器和致动器的控制单元和转换装置。 致动器包括将由控制单元提供的交流电压转换为致动器的有源元件作为尺寸变化。 所述有源元件适于嵌入工具保持器的主体中并且以这种方式使得所述尺寸变化赋予工具保持器的主体弯曲。

公开(公告)号：US06549627B1

公开(公告)日：2003-04-15

申请号：US09016264

申请日：1998-01-30

Applicant: Jim Rasmusson ,  Ingvar Claesson ,  Mattias Dahl ,  Sven Nordholm

Inventor： Jim Rasmusson ,  Ingvar Claesson ,  Mattias Dahl ,  Sven Nordholm

IPC: H04R300

CPC classification number: H04M9/082 ,  G01S7/537 ,  H04R1/406 ,  H04R3/005 ,  H04R2201/401

Abstract: A beamformer is calibrated for use as an acoustic echo canceler in a hands-free communications environment having a loudspeaker and a plurality of microphones. To perform the calibration, a number of adaptive filters are provided in correspondence with each of the microphones, and each of the adaptive filters is trained to model echo properties of the environment as experienced by the corresponding one of the microphones. A target source is activated, thereby generating an acoustic signal that is received by the microphones. The trained adaptive filters are then used to generate jammer signals by, for example, having each one filter a pseudo noise signal. Respective ones of the jammer signals are then combined with corresponding signals supplied by the microphones, thereby generating combination signals. The combination signals are then used to adapt the beamformer to cancel the jammer signals. In another aspect of the invention, the adaptive filters may be utilized during normal operation by having them perform an echo cancellation operation on each of the signals that is to be supplied to the calibrated beamformer.

Abstract translation: 在具有扬声器和多个麦克风的免提通信环境中，波束形成器被校准用作声学回波消除器。 为了执行校准，与每个麦克风相对应地提供多个自适应滤波器，并且训练每个自适应滤波器以对由相应的一个麦克风所经历的环境的回波特性进行建模。 目标源被激活，从而产生由麦克风接收的声信号。 训练后的自适应滤波器然后用于通过例如使每个滤波器具有伪噪声信号来产生干扰信号。 然后将干扰信号中的相应信号与由麦克风提供的相应信号组合，从而产生组合信号。 然后组合信号用于使波束形成器适应以消除干扰信号。 在本发明的另一方面，自适应滤波器可以在正常操作期间通过使自适应滤波器对要提供给校准波束形成器的每个信号执行回波消除操作。

公开(公告)号：US06507623B1

公开(公告)日：2003-01-14

申请号：US09289555

申请日：1999-04-12

Applicant: Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

Inventor： Harald Gustafsson ,  Ingvar Claesson ,  Sven Nordholm

IPC: H04B1500

CPC classification number: G10L21/0208

Abstract: For purposes of noise suppression, spectral subtraction filtering is performed in sample-wise fashion in the time domain using a time-domain representation of a spectral subtraction gain function computed in block-wise fashion in the frequency domain. By continuously performing time-domain filtering on a sample by sample basis, the disclosed methods and apparatus avoid block-processing delays associated with frequency-domain based spectral subtraction systems. Consequently, the disclosed methods and apparatus are particularly well suited for applications requiring very short processing delays. Moreover, since the spectral subtraction gain function is computed in a block-wise fashion in the frequency domain, high quality performance in terms of reduced tonal artifacts and low signal distortion is retained.

Abstract translation: 为了噪声抑制的目的，在时域中以采样方式执行频谱减法滤波，使用在频域中以块方式计算的频谱减法增益函数的时域表示。 通过以样本为基础连续执行时域滤波，所公开的方法和装置避免了与基于频域的谱减法系统相关的块处理延迟。 因此，所公开的方法和装置特别适用于需要非常短的处理延迟的应用。 此外，由于在频域中以块方式计算频谱减法增益函数，所以保留了降低的色调伪影和低信号失真方面的高质量性能。