公开(公告)号：US08559648B2

公开(公告)日：2013-10-15

申请号：US12240523

申请日：2008-09-29

Applicant: Markus Christoph

Inventor： Markus Christoph

IPC: G10K11/175 ,  G10K11/178 ,  G10K11/16

CPC classification number: H04R3/04 ,  H04R2499/13 ,  H04S7/302

Abstract: An active noise cancellation system reduces, at a listening position, the power of a noise signal being radiated from a noise source to the listening position. The system includes an adaptive filter that receives a reference signal representing the noise signal, and provides a compensation signal. A bass management unit receives the compensation signal and applies a phase shift to the compensation signal to provide a phase shifted compensation signal. A first acoustic radiator receives the phase shifted compensation signal and radiates audio indicative thereof to the listening position. A second acoustic radiator receives the compensation signal and radiates audio indicative thereof to the listening position. The transfer function characteristics from the input of the bass management system to the listening position approximately matches a desired transfer function.

Abstract translation: 主动噪声消除系统在收听位置减少从噪声源辐射到收听位置的噪声信号的功率。 该系统包括自适应滤波器，其接收表示噪声信号的参考信号，并提供补偿信号。 低音管理单元接收补偿信号并向补偿信号施加相移以提供相移补偿信号。 第一声​​辐射器接收相移补偿信号并将指示其的音频辐射到收听位置。 第二声辐射器接收补偿信号并将指示其的音频发射到收听位置。 从低音管理系统的输入到收听位置的传递函数特性近似匹配期望的传递函数。

公开(公告)号：US08116481B2

公开(公告)日：2012-02-14

申请号：US11410538

申请日：2006-04-25

Applicant: Markus Christoph

Inventor： Markus Christoph

IPC: A61F11/06 ,  G10K11/16 ,  H03B29/00 ,  H04B15/00

CPC classification number: H03G3/32

Abstract: A system for enhancing the sound signal produced by an audio system in a listening environment by compensating for ambient noise in the listening environment is provided. The system receives an electrical sound signal and generates a sound output therefrom. A total sound signal is sensed representative of the total sound level in the environment, where the total sound level includes both the sound output from the audio system and the ambient noise within the environment. The system extracts an ambient noise signal representative of the ambient noise in the environment from the total sound signal in response to the total sound signal and to a reference signal derived from the electrical sound signal. The system extracts the ambient noise signal using an adaptive filter with an adaptive step size. The system generates a control signal in response to the ambient noise signal and adjusts the sound output of the audio system to compensate for the ambient noise level in response to the control signal. The system calculates a step size for controlling the adaptive step size of the adaptive filter.

Abstract translation: 提供一种用于通过在收听环境中补偿环境噪声来增强听觉环境中的音频系统产生的声音信号的系统。 系统接收电声信号并从中产生声音输出。 感测到的总声音信号代表环境中的总声级，其中总声级包括从音频系统输出的声音和环境中的环境噪声。 该系统响应于总声音信号和从电声信号导出的参考信号，从总声音信号中提取表示环境中的环境噪声的环境噪声信号。 系统使用具有自适应步长的自适应滤波器提取环境噪声信号。 该系统响应于环境噪声信号产生控制信号，并且响应于控制信号调整音频系统的声音输出以补偿环境噪声电平。 系统计算用于控制自适应滤波器的自适应步长的步长。

公开(公告)号：US08009841B2

公开(公告)日：2011-08-30

申请号：US11701629

申请日：2007-02-02

Applicant: Markus Christoph

Inventor： Markus Christoph

IPC: H04R3/00

CPC classification number: H04R3/005 ,  H04R1/406 ,  H04R2201/401 ,  H04R2201/403 ,  H04R2201/405 ,  H04R2430/23 ,  H04R2430/25 ,  H04R2499/13

Abstract: A handsfree communication system includes microphones, a beamformer, and filters. The microphones are spaced apart and are capable of receiving acoustic signals. The beamformer compensates for propagation delays between the direct and reflected acoustic signals. The filters are configured to a predetermined susceptibility level. The filter process the output of the beamformer to enhance the quality of the received signals.

Abstract translation: 免提通信系统包括麦克风，波束形成器和滤波器。 麦克风间隔开并且能够接收声信号。 波束形成器补偿直接和反射声信号之间的传播延迟。 过滤器被配置为预定的敏感度水平。 滤波器处理波束形成器的输出以增强接收信号的质量。

公开(公告)号：US20110206214A1

公开(公告)日：2011-08-25

申请号：US13035393

申请日：2011-02-25

Applicant: Markus Christoph ,  Michael Wurm ,  Michael Perkmann

Inventor： Markus Christoph ,  Michael Wurm ,  Michael Perkmann

IPC: G10K11/16

CPC classification number: G10K11/1784 ,  G10K11/178 ,  G10K2210/1081 ,  G10K2210/3055

Abstract: A system for actively reducing noise at a listening point, includes an earphone housing, a transmitting transducer, a receiving transducer and a controller. The transmitting transducer converts a first electric signal into a first acoustic signal, and radiates the first acoustic signal along a first acoustic path having a first transfer characteristic and along a second acoustic path having a second transfer characteristic. The receiving transducer converts the first acoustic signal and ambient noise into a second electrical signal. The controller compensates for the ambient noise by providing a noise reducing electrical signal to the transmitting transducer. The noise reducing electrical signal is derived from a filtered electrical signal that is provided by filtering the second electrical signal with a third transfer characteristic. The second and the third transfer characteristics together model the first transfer characteristic.

Abstract translation: 用于主动降低收听点噪声的系统包括耳机外壳，发送换能器，接收换能器和控制器。 发送换能器将第一电信号转换成第一声信号，并且沿着具有第一传输特性的第一声路径和沿着具有第二传输特性的第二声路径辐射第一声信号。 接收传感器将第一声信号和环境噪声转换成第二电信号。 控制器通过向发射传感器提供降噪电信号来补偿环境噪声。 噪声降低电信号是从经过滤波的电信号得到的，该电信号通过用第三传输特性滤波第二电信号来提供。 第二和第三传输特性一起模拟了第一传输特性。

公开(公告)号：US07881482B2

公开(公告)日：2011-02-01

申请号：US11434496

申请日：2006-05-15

Applicant: Markus Christoph

Inventor： Markus Christoph

IPC: H03G5/00

CPC classification number: H03H21/0012 ,  H03H21/0014 ,  H03H2021/0041 ,  H03H2021/0092

Abstract: An audio enhancement system is provided for compensating for distortions (e.g., linear distortions) of a sound signal reproduced by an audio system in a listening room. The audio enhancement system includes analysis filters that generate a plurality of analysis output signals from an audio signal to be enhanced. The system also includes synthesis filters that generate an enhanced audio signal from a number of synthesis input signals. The number of analysis output signals and the number of synthesis input signals preferably are equal. Signal processing elements between the analysis filters and the synthesis filters generate one of the synthesis input signals from a respective one of the analysis output signals to perform an inverse filtering for linearizing an unknown transfer function indicative of the audio system and the listening room in the respective frequency range.

Abstract translation: 提供了一种音频增强系统，用于补偿由听音室中的音频系统再现的声音信号的失真（例如，线性失真）。 音频增强系统包括从要增强的音频信号产生多个分析输出信号的分析滤波器。 该系统还包括从多个合成输入信号产生增强音频信号的合成滤波器。 分析输出信号的数量和合成输入信号的数量优选地相等。 分析滤波器和合成滤波器之间的信号处理元件产生来自分析输出信号中的相应一个的合成输入信号之一，以执行逆滤波，以将表示音频系统的未知传输函数和收听室线性化 频率范围。

公开(公告)号：US20100226501A1

公开(公告)日：2010-09-09

申请号：US12718473

申请日：2010-03-05

Applicant: Markus Christoph

Inventor： Markus Christoph

IPC: H04R29/00

CPC classification number: G10L21/0208 ,  G10L21/0216

Abstract: In a system for estimating the power spectral density of acoustical background noise when the level of a smoothed power spectral density signal increases, an increment value is increased, starting from a minimum increment value, by a predetermined amount until a maximum increment value is reached if at the same time the value of the power spectral density currently determined in a new calculation cycle is larger than the estimate value of the power spectral density of the background noise determined in the previous calculation cycle. For cases in which the level of the smoothed power spectral density decreases, the amplitude of the decrement value is increased, starting from a minimum decrement value, by a predetermined amount until a maximum decrement value is reached if at the same time the value of the power spectral density currently determined in a new calculation cycle is smaller than the estimate value of the power spectral density of the background noise determined in the previous calculation cycle.

Abstract translation: 在平滑功率谱密度信号的电平增加时用于估计声背景噪声的功率谱密度的系统中，从最小增量值开始增加预定量的增量值，直到达到最大增量值，如果 同时在新的计算周期中当前确定的功率谱密度的值大于在先前的计算周期中确定的背景噪声的功率谱密度的估计值。 对于平滑功率谱密度的水平降低的情况，减小值的幅度从最小减量值开始增加预定量，直到达到最大减量值，如果同时达到 在新的计算周期中当前确定的功率谱密度小于在先前计算周期中确定的背景噪声的功率谱密度的估计值。

公开(公告)号：US20090314081A1

公开(公告)日：2009-12-24

申请号：US12301367

申请日：2007-05-16

Applicant: Markus Christoph ,  Christian Plankl

Inventor： Markus Christoph ,  Christian Plankl

IPC: G01D11/24

CPC classification number: G01D11/245 ,  G01D11/30

Abstract: A sensor module has a housing that may be mounted about a through opening of an assembly surface. The housing extends at least partly through the through opening when the housing is disposed on the assembly surface. The sensor module further has a sealing body, between the outer surface of the housing and the through opening of the assembly surface and an attachment device, by way of which the housing is fixed to the assembly surface. The attachment device lies on the through opening in order to bring about a mechanical fixation of the housing to the assembly surface.

Abstract translation: 传感器模块具有可围绕组装表面的通孔安装的壳体。 当壳体设置在组件表面上时，壳体至少部分地延伸穿过通孔。 传感器模块还具有密封体，位于壳体的外表面和组件表面的通孔之间，以及附接装置，通过该密封体将壳体固定到组件表面。 附接装置位于通孔上，以便将壳体机械地固定到组件表面。

公开(公告)号：US20090220098A1

公开(公告)日：2009-09-03

申请号：US12396145

申请日：2009-03-02

Applicant: Markus Christoph

Inventor： Markus Christoph

IPC: H04R29/00 ,  H03G5/00

CPC classification number: H04R3/04 ,  H04R2499/13 ,  H04S7/302

Abstract: The invention relates to a method for adapting sound pressure levels in at least one listening location, the sound pressure being generated by a first and a second loudspeaker, each loudspeaker having a supply channel arranged upstream thereto, where at least the supply channel of the second loudspeaker modifies the phase of an audio signal transmitted therethrough according to a phase function. The method includes supplying an audio signal to the supply channels and thus generating an acoustic sound signal; measuring the acoustic sound signal at each listening location and providing corresponding electrical signals representing the measured acoustic sound signal; estimating updated transfer characteristics for each pair of loudspeaker and listening location; calculating an optimum offset phase function based on a mathematical model using the estimated transfer characteristics; updating the phase function by superposing the optimal offset phase function thereto.

Abstract translation: 本发明涉及一种用于调整至少一个收听位置中的声压级的方法，该声压由第一和第二扬声器产生，每个扬声器具有设置在其上游的供应通道，其中至少第二扬声器的供应通道 扬声器根据相位功能修改其中传输的音频信号的相位。 该方法包括：向音频信号提供音频信号，从而产生声音信号; 测量每个收听位置处的声音信号，并提供表示所测量的声音信号的对应电信号; 估计每对扬声器和收听位置的更新传输特性; 基于使用所估计的传送特性的数学模型来计算最佳偏移相位函数; 通过将最佳偏移相位函数叠加来更新相位函数。

公开(公告)号：US20080126461A1

公开(公告)日：2008-05-29

申请号：US11775690

申请日：2007-07-10

Applicant: Markus Christoph

Inventor： Markus Christoph

IPC: G06F17/14

CPC classification number: H04S1/007 ,  H03H17/0213 ,  H04L25/03159 ,  H04L2025/03522

Abstract: The present invention relates to a method for processing a digital input signal by a Finite Impulse Response, FIR, filtering means, comprising partitioning the digital input signal at least partly in the time domain to obtain at least two partitions of the digital input signal; partitioning the FIR filtering means in the time domain to obtain at least two partitions of the FIR filtering means; Fourier transforming each of the at least two partitions of the digital input signal to obtain Fourier transformed signal partitions; Fourier transforming each of the at least two partitions of the FIR filtering means to obtain Fourier transformed filter partitions; performing a convolution of the Fourier transformed signal partitions and the corresponding Fourier transformed filter partitions to obtain spectral partitions; combining the spectral partitions to obtain a total spectrum; and inverse Fourier transforming the total spectrum to obtain a digital output signal.

Abstract translation: 本发明涉及一种通过有限脉冲响应FIR滤波装置处理数字输入信号的方法，包括至少部分地在时域中分割数字输入信号以获得数字输入信号的至少两个分区; 在时域中划分FIR滤波装置以获得FIR滤波装置的至少两个分区; 对数字输入信号的至少两个分区中的每一个进行傅立叶变换以获得傅立叶变换信号分区; 傅里叶变换FIR滤波装置的至少两个分区中的每一个以获得傅立叶变换滤波器分区; 执行傅里叶变换信号分区和对应的傅里叶变换滤波器分区的卷积以获得频谱分区; 组合光谱分区以获得总光谱; 和傅立叶逆变换总谱以获得数字输出信号。

公开(公告)号：US07287050B2

公开(公告)日：2007-10-23

申请号：US10675600

申请日：2003-09-29

Applicant: Markus Christoph

Inventor： Markus Christoph

IPC: G06F17/10

CPC classification number: H03H17/04 ,  H03H17/0294

Abstract: A digital filter includes a delay network with a plurality of delay elements configured and arranged as all-pass filters, having a controllable coefficient value. In the case of a low-pass or high pass filter, the cut-off frequency of the filter can be controlled via the controllable coefficient value associated with phase angle. Similarly, in a bandpass filter, the center frequency is set as a function of the controllable coefficient value.

Abstract translation: 数字滤波器包括延迟网络，其具有被配置和布置为具有可控系数值的全通滤波器的多个延迟元件。 在低通滤波器或高通滤波器的情况下，可以通过与相位角相关的可控系数值来控制滤波器的截止频率。 类似地，在带通滤波器中，将中心频率设置为可控系数值的函数。