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1. 发明授权

US09064502B2 Speech intelligibility predictor and applications thereof 有权
标题翻译：语音清晰度预测及其应用
公开(公告)号：US09064502B2
公开(公告)日：2015-06-23
申请号：US13045303
申请日：2011-03-10
申请人： Cees H. Taal , Richard Hendriks , Richard Heusdens , Ulrik Kjems , Jesper Jensen
发明人： Cees H. Taal , Richard Hendriks , Richard Heusdens , Ulrik Kjems , Jesper Jensen
IPC分类号： G10L21/02 , G10L15/16 , G10L25/69
CPC分类号： G10L25/69
摘要： The application relates to a method of providing a speech intelligibility predictor value for estimating an average listener's ability to understand of a target speech signal when said target speech signal is subject to a processing algorithm and/or is received in a noisy environment. The application further relates to a method of improving a listener's understanding of a target speech signal in a noisy environment and to corresponding device units. The object of the present application is to provide an alternative objective intelligibility measure, e.g. a measure that is suitable for use in a time-frequency environment. The invention may e.g. be used in audio processing systems, e.g. listening systems, e.g. hearing aid systems.
摘要翻译：本申请涉及一种提供语音清晰度预测值的方法，用于当所述目标语音信号经受处理算法和/或在噪声环境中被接收时，估计平均听众理解目标语音信号的能力。该应用还涉及一种改善听众对嘈杂环境中的目标语音信号的理解以及对应的设备单元的方法。本申请的目的是提供替代的目标可懂度度量，例如，适合在时频环境中使用的措施。本发明可以例如用于音频处理系统，例如收听系统，例如助听器系统。

2. 发明申请

US20110224976A1 SPEECH INTELLIGIBILITY PREDICTOR AND APPLICATIONS THEREOF 有权
标题翻译：语音智能预测及其应用
公开(公告)号：US20110224976A1
公开(公告)日：2011-09-15
申请号：US13045303
申请日：2011-03-10
申请人： Cees H. TAAL , Richard Hendriks , Richard Heusdens , Ulrik Kjems , Jesper Jensen
发明人： Cees H. TAAL , Richard Hendriks , Richard Heusdens , Ulrik Kjems , Jesper Jensen
IPC分类号： G10L21/02
CPC分类号： G10L25/69
摘要： The application relates to a method of providing a speech intelligibility predictor value for estimating an average listener's ability to understand of a target speech signal when said target speech signal is subject to a processing algorithm and/or is received in a noisy environment. The application further relates to a method of improving a listener's understanding of a target speech signal in a noisy environment and to corresponding device units. The object of the present application is to provide an alternative objective intelligibility measure, e.g. a measure that is suitable for use in a time-frequency environment. The invention may e.g. be used in audio processing systems, e.g. listening systems, e.g. hearing aid systems.
摘要翻译：本申请涉及一种提供语音清晰度预测值的方法，用于当所述目标语音信号经受处理算法和/或在噪声环境中被接收时，估计平均听众理解目标语音信号的能力。该应用还涉及一种改善听众对嘈杂环境中的目标语音信号的理解以及对应的设备单元的方法。本申请的目的是提供替代的目标可懂度度量，例如，适合在时频环境中使用的措施。本发明可以例如用于音频处理系统，例如收听系统，例如助听器系统。

3. 发明申请

US20100067710A1 NOISE SPECTRUM TRACKING IN NOISY ACOUSTICAL SIGNALS 有权
公开(公告)号：US20100067710A1
公开(公告)日：2010-03-18
申请号：US12550926
申请日：2009-08-31
申请人： Richard C. HENDRIKS , Jesper Jensen , Ulrik Kjems , Richard Heusdens
发明人： Richard C. HENDRIKS , Jesper Jensen , Ulrik Kjems , Richard Heusdens
IPC分类号： H04R29/00
CPC分类号： G10L21/0208 , G10L21/0216 , G10L2021/0575
摘要： The invention relates to a method of estimating noise power spectral density PSD in an input sound signal comprising a noise signal part and a target signal part. The invention further relates to a system to its use. The object of the present invention is to provide a scheme for estimating the noise PSD in an acoustic signal consisting of a target signal contaminated by acoustic noise. The problem is solved by a method comprising the steps of d) providing a digitized electrical input signal to a control path and performing; d1) storing a number of time frames of the input signal each comprising a predefined number N2 of digital time samples xn (n=1, 2, . . . , N2), corresponding to a frame length in time of L2=N2/fs; d2) performing a time to frequency transformation of the stored time frames on a frame by frame basis to provide corresponding spectra Y of frequency samples; d3) deriving a periodogram comprising the energy content |Y|2 for each frequency sample in a spectrum, the energy content being the energy of the sum of the noise and target signal; d4) applying a gain function G to each frequency sample of a spectrum, thereby estimating the noise energy level |Ŵ|2 in each frequency sample, |Ŵ|2=G·|Y|2; d5) dividing the spectra into a number Nsb2 of sub-bands, each sub-band comprising a predetermined number nsb2 of frequency samples, and assuming that the noise PSD level is constant across a sub-band; d6) providing a first estimate |{circumflex over (N)}|2 of the noise PSD level in a sub-band based on the non-zero noise energy levels of the frequency samples in the sub-band; d7) providing a second, improved estimate |Ñ|2 of the noise PSD level in a sub-band by applying a bias compensation factor B to the first estimate, |Ñ|2=B·|{circumflex over (N)}|2. The invention may e.g. be used in listening devices, e.g. hearing aids, mobile telephones, headsets, active earplugs, etc.

4. 发明授权

US08712074B2 Noise spectrum tracking in noisy acoustical signals 有权
标题翻译：噪声频谱跟踪噪声信号
公开(公告)号：US08712074B2
公开(公告)日：2014-04-29
申请号：US12550926
申请日：2009-08-31
申请人： Richard C. Hendriks , Jesper Jensen , Ulrik Kjems , Richard Heusdens
发明人： Richard C. Hendriks , Jesper Jensen , Ulrik Kjems , Richard Heusdens
IPC分类号： H04B15/00
CPC分类号： G10L21/0208 , G10L21/0216 , G10L2021/0575
摘要： A method estimates noise power spectral density (PSD) in an input sound signal to generate an output for noise reduction of the input sound signal. The method includes storing frames of a digitized version of the input signal, each frame having a predefined number N2 of samples corresponding to a frame length in time of L2=N2/sampling frequency. It further includes performing a time to frequency transformation, deriving a periodogram comprising an energy content |Y|2 from the corresponding spectrum Y, applying a gain function G(k,m)=f(σs2(km),σw2l (k,m−1), |Y(k,m)|2), to estimate a noise energy level |Ŵ|2 in each frequency sample, where σs2 is the speech PSD and σw2 the noise PSD. It further includes dividing spectra into a number of sub-bands, and providing a first estimate |{circumflex over (N)}|2 of the noise PSD level in a sub-band and a second, improved estimate |{circumflex over (N)}|2 of the noise PSD level in a subband by applying a bias compensation factor B to the first estimate.
摘要翻译：方法估计输入声音信号中的噪声功率谱密度（PSD）以产生用于输入声音信号的噪声降低的输出。该方法包括存储输入信号的数字化版本的帧，每帧具有对应于L2 = N2 /采样频率的帧长度的预定数量的样本N2。它还包括执行时间到频率变换，从相应的频谱Y导出包括能量含量| Y | 2的周期图，应用增益函数G（k，m）= f（＆sgr; s2（km），＆sgr; w2l （k，m-1），| Y（k，m）| 2），以估计每个频率样本中的噪声能级|Ŵ| 2，其中＆sgr; s2是语音PSD，＆sgr; w2是噪声PSD。它还包括将频谱划分成多个子带，并且提供子带中的噪声PSD电平的第一估计|（在（N）} | 2）和第二估计|改善估计）} | 2，通过将偏置补偿因子B应用于第一估计，子带中的噪声PSD电平。

5. 发明申请

US20070258597A1 Low Frequency Phase Matching for Microphones 审中-公开
标题翻译：麦克风的低频相位匹配
公开(公告)号：US20070258597A1
公开(公告)日：2007-11-08
申请号：US11660734
申请日：2005-08-22
申请人： Karsten Rasmussen , Kim Petersen , Ulrik Kjems
发明人： Karsten Rasmussen , Kim Petersen , Ulrik Kjems
IPC分类号： H04R5/00
CPC分类号： H04R29/006
摘要： The invention relates to a communication device having at least two microphones, where in order to match the microphone performance in respect of the phase response a correction filter in the form of a IIR filter is implemented and where the amplitude of the transfer function for the correction filter is the inverse of the difference between the two microphone amplitudes.
摘要翻译：本发明涉及具有至少两个麦克风的通信设备，其中为了匹配相对于相位响应的麦克风性能，实现了IIR滤波器形式的校正滤波器，并且其中用于校正的传递函数的幅度滤波器是两个麦克风幅度之差的倒数。

6. 发明申请

US20110051948A1 METHOD OF CORRECTING ERRORS IN BINARY MASKS 有权
标题翻译：校正二进制面板中的错误的方法
公开(公告)号：US20110051948A1
公开(公告)日：2011-03-03
申请号：US12850439
申请日：2010-08-04
申请人： Jesper Bünsow BOLDT , Ulrik Kjems , Michael Syskind Pedersen , Mads Graesbøll Christensen , Søren Holdt Jensen
发明人： Jesper Bünsow BOLDT , Ulrik Kjems , Michael Syskind Pedersen , Mads Graesbøll Christensen , Søren Holdt Jensen
IPC分类号： H04R3/02
CPC分类号： H04R25/505 , G10L15/142 , G10L21/02 , G10L25/18 , G10L25/48 , G10L2021/065 , H04R2225/43
摘要： The invention relates to a method of identifying and correcting errors in a noisy binary mask. An object of the present invention is to provide a scheme for improving a binary mask representing speech. The problem is solved in that the method comprises a) providing a noisy binary mask comprising a binary representation of the power density of an acoustic signal comprising a target signal and a noise signal at a predefined number of discrete frequencies and a number of discrete time instances; b) providing a statistical model of a clean binary mask representing the power density of the target signal; and c) using the statistical model to detect and correct errors in the noisy binary mask. This has the advantage of providing an alternative and relatively simple way of improving an estimate of a binary mask representing a speech signal. The invention may e.g. be used for speech processing, e.g. in a hearing instrument.
摘要翻译：本发明涉及一种识别和纠正嘈杂二进制掩码中的错误的方法。本发明的目的是提供一种用于改进表示语音的二进制掩码的方案。解决的问题是，该方法包括：a）提供噪声二进制掩模，其包括包含目标信号的声信号的功率密度和预定数量的离散频率的噪声信号和多个离散时间实例的二进制表示 ; b）提供表示目标信号的功率密度的干净的二进制掩码的统计模型; 和c）使用统计模型来检测和纠正噪声二进制掩码中的错误。这具有提供改进代表语音信号的二进制掩码的估计的替代且相对简单的方法的优点。本发明可以例如用于语音处理，例如。在听力仪器中。

7. 发明授权

US08626495B2 Method of correcting errors in binary masks 有权
标题翻译：纠正二进制掩码错误的方法
公开(公告)号：US08626495B2
公开(公告)日：2014-01-07
申请号：US12850439
申请日：2010-08-04
申请人： Jesper Bünsow Boldt , Ulrik Kjems , Michael Syskind Pedersen , Mads Graesbøll Christensen , Søren Holdt Jensen
发明人： Jesper Bünsow Boldt , Ulrik Kjems , Michael Syskind Pedersen , Mads Graesbøll Christensen , Søren Holdt Jensen
IPC分类号： G10L19/00
CPC分类号： H04R25/505 , G10L15/142 , G10L21/02 , G10L25/18 , G10L25/48 , G10L2021/065 , H04R2225/43
摘要： The invention relates to a method of identifying and correcting errors in a noisy binary mask. An object of the present invention is to provide a scheme for improving a binary mask representing speech. The problem is solved in that the method comprises a) providing a noisy binary mask comprising a binary representation of the power density of an acoustic signal comprising a target signal and a noise signal at a predefined number of discrete frequencies and a number of discrete time instances; b) providing a statistical model of a clean binary mask representing the power density of the target signal; and c) using the statistical model to detect and correct errors in the noisy binary mask. This has the advantage of providing an alternative and relatively simple way of improving an estimate of a binary mask representing a speech signal. The invention may e.g. be used for speech processing, e.g. in a hearing instrument.
摘要翻译：本发明涉及一种识别和纠正嘈杂二进制掩码中的错误的方法。本发明的目的是提供一种用于改进表示语音的二进制掩码的方案。解决的问题是，该方法包括：a）提供噪声二进制掩模，其包括包含目标信号的声信号的功率密度和预定数量的离散频率的噪声信号和多个离散时间实例的二进制表示 ; b）提供表示目标信号的功率密度的干净的二进制掩码的统计模型; 和c）使用统计模型来检测和纠正噪声二进制掩码中的错误。这具有提供改进代表语音信号的二进制掩码的估计的替代且相对简单的方法的优点。本发明可以例如用于语音处理，例如。在听力仪器中。

8. 发明申请

US20100128911A1 METHOD TO REDUCE FEEDBACK IN HEARING AIDS 有权
标题翻译：减少听觉艾滋病反馈的方法
公开(公告)号：US20100128911A1
公开(公告)日：2010-05-27
申请号：US12624088
申请日：2009-11-23
申请人： Thomas Bo Elmedyb , Michael Syskind Pedersen , Ulrik Kjems , Thomas Kaulberg
发明人： Thomas Bo Elmedyb , Michael Syskind Pedersen , Ulrik Kjems , Thomas Kaulberg
IPC分类号： H04R25/00
CPC分类号： H04R25/453 , H04R25/558 , H04R2430/03
摘要： Disclosed is a method of reducing feedback in a hearing aid adapted to be worn by a user, the method comprising the step of: receiving an audio input signal in an input transducer in the hearing aid; wherein the method further comprises the steps of: transforming the input signal into the frequency domain; dividing the audio signal into a plurality of frequency bands; determining a threshold frequency over which a plurality of upper frequency bands lies; multiplying each of the plurality of upper frequency bands by a random phase, thereby obtaining a plurality of phase randomized upper frequency bands; synthesizing the plurality of phase randomized upper frequency bands and the lower frequency bands to an output signal; transforming the output signal into the time-domain; and transmitting the output signal to an output transducer of the hearing aid.
摘要翻译：公开了一种降低助听器中的反馈的方法，适用于用户佩戴，该方法包括以下步骤：在助听器中的输入换能器中接收音频输入信号; 其中所述方法还包括以下步骤：将所述输入信号变换成频域; 将音频信号划分成多个频带; 确定多个上频带所在的阈值频率; 将多个高频带中的每一个乘以随机相位，从而获得多个相位随机化的高频带; 将多个相位随机化的高频带和较低频带合成为输出信号; 将输出信号转换为时域; 以及将所述输出信号发送到所述助听器的输出换能器。

9. 发明授权

US07885991B2 Digital filter having a fir filter and a warped fir filter, and a listening device including such a digital filter 有权
标题翻译：具有冷杉滤光器和翘曲冷杉滤光器的数字滤光器，以及包括这种数字滤波器的收听装置
公开(公告)号：US07885991B2
公开(公告)日：2011-02-08
申请号：US10547687
申请日：2004-03-03
申请人： Thomas Behrens , Ulrik Kjems , Thomas Bo Elmedyb
发明人： Thomas Behrens , Ulrik Kjems , Thomas Bo Elmedyb
IPC分类号： G06F17/10
CPC分类号： H03H17/0261 , H03H17/0286 , H03H17/06
摘要： A digital filter includes at least a first and a second delayed summation line. One of the delay lines includes a warped finite impulse response (FIR) filter and the other line is a FIR filter. Midpoint elements from the first delayed summation line are used as input to the second delayed summation line. Output from the first delayed summation line is delayed with a delay corresponding to the total delay of the second delayed summation line and the delayed output from the first delayed summation line is added to output from the second delayed summation line to faun a new output.
摘要翻译：数字滤波器包括至少第一和第二延迟求和线。延迟线之一包括翘曲有限脉冲响应（FIR）滤波器，另一行是FIR滤波器。来自第一延迟求和线的中点元素被用作第二延迟求和线的输入。来自第一延迟求和线的输出以对应于第二延迟求和线的总延迟的延迟延迟，并且来自第一延迟求和线的延迟输出被加到从第二延迟求和线输出以产生新的输出。

10. 发明授权

US07340068B2 Device and method for detecting wind noise 有权
标题翻译：风噪声检测装置及方法
公开(公告)号：US07340068B2
公开(公告)日：2008-03-04
申请号：US10367955
申请日：2003-02-19
申请人： Kim Spetzler Petersen , Gudmundur Bogason , Ulrik Kjems , Thomas Bo Elmedyb
发明人： Kim Spetzler Petersen , Gudmundur Bogason , Ulrik Kjems , Thomas Bo Elmedyb
IPC分类号： H04R3/00 , H04R25/00 , H04B15/00
CPC分类号： H04R3/005 , G02B6/4214 , H04R25/407 , H04R2410/07
摘要： A device for detecting the presence of wind noise in an array of microphones including two or more separate sound inlet openings, and a sound to electrical converting element or microphone in relation to each sound inlet opening. The microphones each generate time dependent signals which are fed to a signal processing device that provides one or more output signals. The signal processing device has means for generating a time dependent cross correlation function between a first and a second microphone signal, and means for generating a signal corresponding to a time dependent auto correlation function of either the first or the second of the microphone signals. Further, the signal processing device has means for comparing the values of the auto correlation function and the cross correlation function and the means for comparing are arranged to detect the condition that the auto correlation function is substantially higher than the cross correlation function, whereby the condition is indicative of the presence of wind noise.
摘要翻译：一种用于检测麦克风阵列中存在的风噪声的装置，包括两个或更多个独立的声音入口开口，以及相对于每个声音入口开口的声音到电转换元件或麦克风。麦克风各自产生时间相关信号，其被馈送到提供一个或多个输出信号的信号处理装置。信号处理装置具有用于在第一麦克风信号和第二麦克风信号之间产生与时间相关的互相关函数的装置，以及用于产生对应于麦克风信号的第一或第二麦克风信号的与时间相关的自相关函数的信号的装置。此外，信号处理装置具有用于比较自相关函数和互相关函数的值的装置，并且用于比较的装置被设置为检测自相关函数显着高于互相关函数的条件，由此条件表示风噪声的存在。

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