公开(公告)号：US08107321B2

公开(公告)日：2012-01-31

申请号：US12602640

申请日：2007-06-01

Applicant: Marián Képesi ,  Michael Wohlmayr ,  Gernot Kubin

Inventor： Marián Képesi ,  Michael Wohlmayr ,  Gernot Kubin

IPC: G01S3/80

CPC classification number: G01S3/786 ,  G10L25/90 ,  G10L2021/02166

Abstract: The invention relates to a method for localizing and tracking acoustic sources (101) in a multi-source environment, comprising the steps of recording audio-signals (103) of at least one acoustic source (101) with at least two recording means (104, 105), creating a two- or multi-channel recording signal, partitioning said recording signal into frames of predefined length (N), calculating for each frame a cross-correlation function as a function of discrete time-lag values (τ) for channel pairs (106, 107) of the recording signal, evaluating the cross-correlation function by calculating a sampling function depending on a pitch parameter (f0) and at least one spatial parameter (φ0), the sampling function assigning a value to every point of a multidimensional space being spanned by the pitch-parameter and the spatial parameters, and identifying peaks in said multidimensional space with respective acoustic sources in the multi-source environment.

Abstract translation: 本发明涉及一种用于在多源环境中定位和跟踪声源（101）的方法，包括以下步骤：用至少两个记录装置（104）记录至少一个声源（101）的音频信号（103） ，105），创建两通道或多通道记录信号，将所述记录信号划分成预定长度（N）的帧，为每帧计算作为离散时间滞后值（τ）的函数的互相关函数 信道对（106,107），通过计算取决于音调参数（f0）和至少一个空间参数（＆phgr; 0）的采样函数来评估互相关函数，该采样函数将值分配给 通过音调参数和空间参数跨越多维空间的每个点，以及在多源环境中用相应声源识别所述多​​维空间中的峰值。

公开(公告)号：US20050286668A1

公开(公告)日：2005-12-29

申请号：US11133119

申请日：2005-05-19

Applicant: Gernot Kubin ,  Gerhard Paoli ,  David Schwingshackl

Inventor： Gernot Kubin ,  Gerhard Paoli ,  David Schwingshackl

IPC: H03H17/02 ,  H04B3/23 ,  H04L7/00 ,  H04L25/03

CPC classification number: H03H17/0223 ,  H03H17/0261 ,  H03H17/0275 ,  H03H17/0685

Abstract: A method for creating a form of a non-linear filter suitable for reducing a computational complexity is proposed. The filter is resolved into polyphase components in such a way that the polyphase components can be interchanged with a conversion of the sampling rate of a signal to be sent to the filter or of a signal to be emitted by the filter. Corresponding filters and filter arrangements are also proposed. In this way, a computational complexity for calculating the signal to be emitted by the filter can be significantly simplified. The invention can be used in echo compensation.

Abstract translation: 提出了一种用于创建适合于降低计算复杂度的非线性滤波器形式的方法。 滤波器被解析为多相分量，使得多相分量可以与要发送到滤波器的信号的采样速率或由滤波器发射的信号的变换相互交换。 还提出了相应的滤波器和滤波器布置。 以这种方式，可以显着简化用于计算由滤波器发射的信号的计算复杂度。 本发明可用于回波补偿。

公开(公告)号：US20120004906A1

公开(公告)日：2012-01-05

申请号：US13147893

申请日：2010-02-01

Applicant: Martin Hagmuller ,  Gernot Kubin

Inventor： Martin Hagmuller ,  Gernot Kubin

IPC: G10L21/02

CPC classification number: G10L21/0364 ,  G10L25/03 ,  G10L25/18

Abstract: In order to improve the speech quality of an electric larynx (EL) speaker, the speech signal of which is digitized by suitable means, the following steps are carried out: a) dividing a single-channel speech signal into a series of frequency channels by transferring it from a time domain into a discrete frequence domain; b) filtering out the modulation frequency of the EL by way of a high-pass or notch filter, in each frequency channel; and c) back-transforming the filtered speech signal from the frequency domain into the time domain and combining it into a single-channel output signal.

Abstract translation: 为了提高电话喉音（EL）扬声器的语音质量，其语音信号通过适当的方式数字化，执行以下步骤：a）将单声道语音信号划分为一系列频道，由 将其从时域转移到离散频域; b）在每个频道中通过高通或陷波滤波器滤除EL的调制频率; 以及c）将经滤波的语音信号从频域逆变换到时域并将其组合成单通道输出信号。

公开(公告)号：US20100049507A1

公开(公告)日：2010-02-25

申请号：US12440952

申请日：2007-09-06

Applicant: Erhard Rank ,  Gernot Kubin

Inventor： Erhard Rank ,  Gernot Kubin

IPC: G10L21/02 ,  G10L19/04

CPC classification number: G10L21/0208 ,  G10L25/12

Abstract: An apparatus for noise suppression having a linear prediction analysis circuit having an LP error filter (LFF), which takes a first, noisy voice signal y(n)=x(n)+ε(n) as a basis for producing an LP-error-filter output signal e(n), having a coefficient calculation unit (KBE), which updates the coefficients of the LP error filter on the basis of the internal signals (including the input and out signals y(n) and e(n)) in the LP error filter, and having a subtraction unit, which subtracts the LP error filter output signal e(n) from the first voice signal y(n) in a subtractor and, following the subtraction, outputs the remainder as a second voice signal x(n)=y(n)−e(n) in which the noise is suppressed. A noise estimation unit (GSE) is provided which takes the internal signals of the LP error filter as a basis for producing a noise power signal σn2 and a voice power signal σx2, these signals are applied to the coefficient calculation unit (KBE) and said signals are used by the latter for the purpose of optimizing the noise suppression.

Abstract translation: 一种用于噪声抑制的装置，具有具有LP误差滤波器（LFF）的线性预测分析电路，其采用第一噪声语音信号y（n）= x（n）+ egr（n）作为产生LP 具有系数计算单元（KBE）的误差滤波器输出信号e（n），其基于内部信号（包括输入和输出信号y（n）和e（n））更新LP误差滤波器的系数， n）），并且具有减法单元，该减法单元在减法器中从第一语音信号y（n）中减去LP误差滤波器输出信号e（n），并且在减法之后将剩余部分输出为 第二语音信号x（n）= y（n）-e（n），噪声被抑制。 提供了一种噪声估计单元（GSE），其将LP误差滤波器的内部信号作为产生噪声功率信号sgr n2和语音功率信号sgr x2的基础，将这些信号施加到系数计算单元 KBE），并且所述信号由后者用于优化噪声抑制的目的。

公开(公告)号：US07501967B2

公开(公告)日：2009-03-10

申请号：US11244569

申请日：2005-10-06

Applicant: Dieter Draxelmayr ,  Gernot Kubin ,  Christian Vogel

Inventor： Dieter Draxelmayr ,  Gernot Kubin ,  Christian Vogel

IPC: H03M1/00

CPC classification number: H03M1/1215

Abstract: An arrangement for a time interleaved analog-to-digital converter that converts an signal to a digital signal and has a converter array with a plurality of analog-to-digital converters arranged in a fixed sequence in parallel with one another and can be operated with staggered timing with respect to one another is disclosed. The arrangement has a connection network which, for the purposes of actuation with staggered timing, generates in each case one control signal for an individual analog-to-digital converter in each case, with the connection network predefining the time sequence with which the control signals actuate the individual analog-to-digital converters in such a way that owing to this sequence of the control signals and thus the sequence of the actuated individual analog-to-digital converters there is at least a reduction in an interference spectrum in the spectrum of the input and/or output signal. A sorting method for operating this analog-to-digital converter is also disclosed.

Abstract translation: 一种用于时间交织的模数转换器的装置，其将信号转换为数字信号，并且具有转换器阵列，所述转换器阵列具有以彼此并联的固定顺序排列的多个模数转换器并且可以与 披露了相互交错的时序。 该装置具有连接网络，为了以交错的定时进行动作，在每种情况下，在每种情况下都产生用于各个模拟 - 数字转换器的一个控制信号，其中连接网络预先定义控制信号 以这样的方式致动各个模拟 - 数字转换器，即由于控制信号的这一序列以及由此驱动的各个模 - 数转换器的序列至少降低了频谱中的干扰谱 输入和/或输出信号。 还公开了用于操作该模数转换器的排序方法。

公开(公告)号：US20060097901A1

公开(公告)日：2006-05-11

申请号：US11244569

申请日：2005-10-06

Applicant: Dieter Draxelmayr ,  Gernot Kubin ,  Christian Vogel

Inventor： Dieter Draxelmayr ,  Gernot Kubin ,  Christian Vogel

IPC: H03M1/12

CPC classification number: H03M1/1215

Abstract: An arrangement for a time interleaved analog-to-digital converter that converts an signal to a digital signal and has a converter array with a plurality of analog-to-digital converters arranged in a fixed sequence in parallel with one another and can be operated with staggered timing with respect to one another is disclosed. The arrangement has a connection network which, for the purposes of actuation with staggered timing, generates in each case one control signal for an individual analog-to-digital converter in each case, with the connection network predefining the time sequence with which the control signals actuate the individual analog-to-digital converters in such a way that owing to this sequence of the control signals and thus the sequence of the actuated individual analog-to-digital converters there is at least a reduction in an interference spectrum in the spectrum of the input and/or output signal. A sorting method for operating this analog-to-digital converter is also disclosed.