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    • 3. 发明公开
    • Method to reduce feedback in hearing aids
    • 在助听器中,以及相应的装置和相应的计算机程序产品,用于减少反馈的方法
    • EP2190217A1
    • 2010-05-26
    • EP08105855.4
    • 2008-11-24
    • Oticon A/S
    • Pedersen, Michael SyskindElmedyb, Thomas BoKjems, UlrikKaulberg, Thomas
    • H04R25/00
    • H04R25/453H04R25/558H04R2430/03
    • 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.
    • 公开的是由用户佩戴助听器angepasst减少反馈的方法,该方法包括以下步骤:在音频输入信号中接收在所述助听器的输入变换器; worin该方法还包括如下步骤:将输入信号变换到频域; 将音频信号划分成频带的多元性; 确定性采矿阈值频率在其上部频带中的多元性所在; 由随机相位相乘的每个上部频带中的多个,由此获得的相位的多元性随机化的上部频带; 合成的相位多元性随机化的上部频带和下部频带以输出信号; 将所述输出信号变换到时域; 和所述输出信号传送到输出到换能器的助听器的。
    • 5. 发明公开
    • Method to reduce feed-back in hearing aids
    • 赫尔格堡的维尔法罕zurRückmeldungsreduktion
    • EP2442590A2
    • 2012-04-18
    • EP12150551.5
    • 2008-11-24
    • Oticon A/S
    • Elmedyb, Thomas BoPedersen, Michael SyskindKjems, UlrikKaulberg, Thomas
    • H04R25/00
    • H04R25/453H04R25/558H04R2430/03
    • Disclosed is a method of reducing feedback in a hearing aid system comprising left and right hearing aids, each hearing aid being adapted to be worn by a user and for communicating with each other, 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, wherein the same random phase is changed by the same amount in the left and the right hearing aids for each upper frequency band. A hearing aid system comprising left and right hearing aids adapted to communicate with each other is further more disclosed.
    • 公开了一种减少助听器系统中的反馈的方法,该助听器系统包括左和右助听器,每个助听器适于由用户佩戴并且用于彼此通信,所述方法包括以下步骤:接收音频输入信号 助听器中的输入换能器; 其中所述方法还包括以下步骤:将所述输入信号变换成所述频域; 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,其中在每个上部频带的左侧和右侧助听器中相同的随机相位改变相同的量。 进一步公开了包括适于彼此通信的左右助听器的助听器系统。
    • 7. 发明公开
    • A method of correcting errors in binary masks
    • 弗拉赫伦·祖尔·费勒克雷克图尔
    • EP2306449A1
    • 2011-04-06
    • EP09168699.8
    • 2009-08-26
    • OTICON A/S
    • Boldt, Jesper BünsowKjems, UlrikPedersen, Michael SyskindChristensen, Mads GræsbøllJensen, Søren Holdt
    • G10L11/00
    • H04R25/505G10L15/142G10L21/02G10L25/18G10L25/48G10L2021/065H04R2225/43
    • 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 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 the speech processing, e.g. in a hearing instrument.
    • 本发明涉及一种识别和纠正噪声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和噪声信号的声信号的功率密度的二进制表示,所述噪声信号以预定数量的离散频率和多个离散时间实例 ; b)提供表示目标信号的干净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。 这具有提供改进对表示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. 发明公开
    • Method of estimating weighting function of audio signals in a hearing aid
    • 在einemHörgerät的Verfahren zurSchätzungder Gewichtungsfunktion von Audiosignalen
    • EP2088802A1
    • 2009-08-12
    • EP08101366.6
    • 2008-02-07
    • OTICON A/S
    • Elmedyb, Thomas BoRasmussen, Karsten BoKjems, UlrikPedersen, Michael SyskindBoldt, Jesper Bünsow
    • H04R25/00H04R1/40H04R3/00
    • H04R25/407H04R1/406H04R25/453H04R2225/021H04S2420/01
    • Disclosed is method of generating an audible signal in a hearing aid by estimating a weighting function of received audio signals, the hearing aid is adapted to be worn by a user; the method comprises the steps of:
      estimating a directional signal by estimating a weighted sum of two or more microphone signals from two or more microphones, where a first microphone of the two or more microphones is a front microphone, and where a second microphone of the two or more microphones is a rear microphone;
      estimating a direction-dependent time-frequency gain, and
      synthesizing an output signal;

      wherein estimating the direction-dependent time-frequency gain comprises:
      • obtaining at least two directional signals each containing a time-frequency representation of a target signal and a noise signal; and where a first of the directional signals is defined as a front aiming signal, and where a second of the directional signals is defined as a rear aiming signal;
      • using the time-frequency representation of the target signal and the noise signal to estimate a time-frequency mask; and
      • using the estimated time-frequency mask to estimate the direction-dependent time-frequency gain.
    • 公开了通过估计接收到的音频信号的加权函数在助听器中产生听觉信号的方法,助听器适于由用户佩戴; 该方法包括以下步骤:通过估计来自两个或更多个麦克风的两个或多个麦克风信号的加权和估计方向信号,其中两个或更多麦克风的第一麦克风是前麦克风,并且其中第二麦克风 两个或更多麦克风是后麦克风; 估计与方向有关的时频增益,并合成输出信号; 其中估计所述取决于方向的时间 - 频率增益包括:获得每个包含目标信号和噪声信号的时间频率表示的至少两个方向信号; 并且其中第一定向信号被定义为前瞄准信号,并且其中第二方向信号被定义为后瞄准信号; 使用目标信号的时间频率表示和噪声信号来估计时间频率掩模; 并使用估计的时频掩模来估计方向相关的时间 - 频率增益。
    • 10. 发明公开
    • Noise estimation for use with noise reduction and echo cancellation in personal communication
    • Geräuschschätzungzur Verwendung mitGeräuschreduzierungundEchounterdrückunginpersönlicherKommunikation
    • EP2701145A1
    • 2014-02-26
    • EP12181723.3
    • 2012-08-24
    • Retune DSP ApSOTICON A/S
    • Kjems, UlrikJensen, Jesper
    • G10L21/0232G10L21/0216H04R3/00
    • G10L15/20G10L21/0232G10L2021/02166H04M9/082H04R1/1083H04R3/005H04R3/02H04R2225/43H04R2227/009
    • The application relates to a method for audio signal processing. The application further relates to a method of processing signals obtained from a multi-microphone system. The object of the present application is to reduce undesired noise sources and residual echo signals from an initial echo cancellation step. The problem is solved by
      receiving M communication signals in frequency subbands where M is at least two;
      processing the M subband communication signals in each subband with a blocking matrix ( 203,303,403 ) of M rows and N linearly independent columns in each subband, where N >=1 and N M , to obtain N target-cancelled signals in each subband;
      processing the M subband communication signals and the N target-cancelled signals in each subband with a set of beamformer coefficients ( 204,304,404 ) to obtain a beamformer output signal in each subband;
      processing the communication signals with a target absence detector ( 309 ) to obtain a target absence signal in each subband;
      using the target absence signal to obtain an inverse target-cancelled covariance matrix of order N ( 310,410 ) in each band;
      processing the N target-cancelled signals in each subband with the inverse target-cancelled covariance matrix in a quadratic form ( 312, 412 ) to yield a real-valued noise correction factor in each subband;
      using the target absence signal to obtain an initial estimate ( 311, 411 ) of the noise power in the beamformer output signal averaged over recent frames with target absence in each subband;
      multiplying the initial noise estimate with the noise correction factor to obtain a refined estimate ( 417 ) of the power of the beamformer output noise signal component in each subband;
      processing the refined estimate of the power of the beamformer output noise signal component with the magnitude of the beamformer output to obtain a postfilter gain value in each subband;
      processing the beamformer output signal with the postfilter gain value ( 206,306,406 ) to obtain a postfilter output signal in each subband;
      processing the postfilter output subband signals through a synthesis filterbank ( 207,307,407 ) to obtain an enhanced beamformed output signal where the target signal is enhanced by attenuation of noise signal components. This has the advantage of providing improved sound quality and reduction of undesired signal components such as the late reverberant part of an acoustic echo signal. The invention may e.g. be used for headsets, hearing aids, active ear protection systems, mobile telephones, teleconferencing systems, karaoke systems, public address systems, mobile communication devices, hands-free communication devices, voice control systems, car audio systems, navigation systems, audio capture, video cameras, and video telephony.
    • 本申请涉及音频信号处理方法。 该应用还涉及一种处理从多麦克风系统获得的信号的方法。 本申请的目的是从初始回波消除步骤减少不期望的噪声源和残留回波信号。 该问题通过在M至少为2的频率子带中接收M个通信信号来解决; 在每个子带中以每个子带中的M行和N个线性独立列的阻塞矩阵(203,303,403)处理每个子带中的M个子带通信信号,其中N> = 1和N