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1. 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。

2. 发明授权

US08542855B2 System for reducing acoustic feedback in hearing aids using inter-aural signal transmission, method and use 有权
标题翻译：使用听觉信号传输，方法和用途减少助听器中的声反馈的系统
公开(公告)号：US08542855B2
公开(公告)日：2013-09-24
申请号：US12506635
申请日：2009-07-21
申请人： Thomas Bo Elmedyb , Jesper Jensen , Meng Guo
发明人： Thomas Bo Elmedyb , Jesper Jensen , Meng Guo
IPC分类号： H04R25/00
CPC分类号： H04R25/552 , H04R5/033 , H04R25/453
摘要： The invention relates to a hearing aid system comprising first and second spatially separated hearing instruments, the system being adapted for processing input sounds to output sounds according to a user's needs. The invention further relates to a method and use. The object of the present invention is to provide an alternative scheme for reducing the effect of acoustic feedback in a hearing aid system. The problem is solved in that the hearing instruments comprises, respectively, first and second input transducers for converting a first input sound to first and second electric input signal, and first and second output transducers for converting first and second processed electric output signal to first and second output sounds, wherein the system is adapted to provide that a first Tx-signal originating from the first electric input signal of the first hearing instrument is transmitted to the second hearing instrument and used in the formation of the second processed electric output signal, and that a second Tx-signal originating from the second electric input signal of the second hearing instrument is transmitted to the first hearing instrument and used in the formation of the first processed electric output signal. This has the advantage of providing a scheme for reducing or effectively eliminating acoustic feedback in a pair of hearing instruments. The invention may e.g. be used in listening devices, e.g. hearing aids, head sets, or active ear plugs.
摘要翻译：本发明涉及一种包括第一和第二空间分离的听力仪器的助听器系统，该系统适于根据用户的需要处理输入声音以输出声音。本发明还涉及一种方法和用途。本发明的目的是提供一种用于减少助听器系统中声反馈的影响的替代方案。解决的问题在于，听力仪器分别包括用于将第一输入声音转换为第一和第二电输入信号的第一和第二输入换能器，以及用于将第一和第二处理电输出信号转换为第一和第二输入信号的第一和第二输出换能器第二输出声音，其中所述系统适于提供源自所述第一听力仪器的第一电输入信号的第一Tx信号被发送到所述第二听力仪器并用于形成所述第二处理电输出信号，以及来自第二听力仪器的第二电输入信号的第二Tx信号被发送到第一听力仪器，并用于形成第一处理电输出信号。这具有提供用于减少或有效地消除一对听力仪器中的声反馈的方案的优点。本发明可以例如用于听音装置，例如助听器，头戴式耳塞或有源耳塞。

3. 发明授权

US08422708B2 Adaptive long-term prediction filter for adaptive whitening 有权
标题翻译：用于自适应白化的自适应长期预测滤波器
公开(公告)号：US08422708B2
公开(公告)日：2013-04-16
申请号：US12506983
申请日：2009-07-21
申请人： Thomas Bo Elmedyb , Jesper Jensen
发明人： Thomas Bo Elmedyb , Jesper Jensen
IPC分类号： H04R25/00 , H04B15/00
CPC分类号： H04R25/453 , H04R5/033
摘要： A method of estimating acoustic feedback in a hearing instrument in order to reduce the impact of tonal components of acoustic feedback. The hearing instrument comprises an input transducer, an output transducer, a forward path being defined between the input transducer and the output transducer, a signal processing unit defining an input side and an output side of the forward path, and a feedback loop from the output side to the input side. The feedback loop comprises a feedback path estimation unit receiving first and second estimation input signals from the input and output side of the forward path, respectively, wherein the first and second estimation input signal paths comprise first and second long term prediction filters P(z), the feedback cancellation system being adapted to provide that the variable parameters of at least one of the long term prediction filters are estimated based on the filter input signal.
摘要翻译：一种在听力仪器中估计声反馈以减少声反馈的音调分量的影响的方法。听力仪器包括输入换能器，输出换能器，在输入换能器和输出换能器之间定义的前进路径，定义前进路径的输入侧和输出侧的信号处理单元，以及来自输出的反馈回路侧到输入端。反馈回路包括反馈路径估计单元，分别从前向通路的输入和输出侧接收第一和第二估计输入信号，其中第一和第二估计输入信号路径包括第一和第二长期预测滤波器P（z）反馈消除系统适于提供基于滤波器输入信号来估计长期预测滤波器中的至少一个的可变参数。

4. 发明授权

US08422707B2 Spectral content modification for robust feedback channel estimation 有权
标题翻译：用于鲁棒反馈信道估计的频谱内容修改
公开(公告)号：US08422707B2
公开(公告)日：2013-04-16
申请号：US12506424
申请日：2009-07-21
申请人： Thomas Bo Elmedyb , Jesper Jensen
发明人： Thomas Bo Elmedyb , Jesper Jensen
IPC分类号： H04R25/00 , H04B15/00
CPC分类号： H04R25/353 , H04R25/453
摘要： A listening device includes an input and output transducers and a forward path defined between the transducers. The forward path may include a signal processing unit processing an SPU-input signal originating from the electric input signal in a time-frequency representation having successive time frames each having a frequency spectrum of the signal in the time frame in question. The signal processing unit includes a spectral content modification unit to modify values of the signal of one or more regions of the frequency spectrum of a given time frame so that the modified values are less correlated to the corresponding time-frequency regions of the input signal. A feedback path estimation unit is uses he improved processed output signal in the feedback estimation. The spectral content modification unit may base the modification on a model of the human auditory system so that the modifications are not perceptible by the user.
摘要翻译：聆听装置包括输入和输出换能器以及在换能器之间限定的前进路径。前向路径可以包括信号处理单元，其处理来自电输入信号的SPU输入信号，时间频率表示具有连续的时间帧，每个时间帧在所讨论的时间帧中具有该信号的频谱。信号处理单元包括频谱内容修改单元，用于修改给定时间帧的频谱的一个或多个区域的信号的值，使得修改值与输入信号的对应的时间 - 频率区域相关性较小。反馈路径估计单元在反馈估计中使用他改进的经处理的输出信号。光谱内容修改单元可以将修改基于人类听觉系统的模型，使得修改不被用户感知。

5. 发明授权

US08295519B2 Codebook based feedback path estimation 有权
标题翻译：基于码本的反馈路径估计
公开(公告)号：US08295519B2
公开(公告)日：2012-10-23
申请号：US12506793
申请日：2009-07-21
申请人： Thomas Bo Elmedyb , Jesper Jensen
发明人： Thomas Bo Elmedyb , Jesper Jensen
IPC分类号： H04R25/00
CPC分类号： H04R25/453 , H04R25/70
摘要： The invention relates to a hearing instrument for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of operating a hearing instrument and to use of a hearing instrument. The object of the present invention is to provide an alternative scheme for handling acoustic feedback in a hearing instrument. The problem is solved in that an input transducer for converting an input sound to an electric input signal and an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer, a feedback cancellation system for estimating the effect of acoustic feedback from the output transducer to the input transducer, the feedback cancellation system comprising a variable pre-estimated filter and a memory wherein a number of predetermined feedback channel impulse responses corresponding to a number of acoustic environments where substantial feedback is experienced are stored, and wherein the hearing instrument comprises a monitoring unit that—based on the current acoustic environment—is adapted to choose the currently most appropriate impulse response of the variable pre-estimated filter among the stored impulse responses. This has the advantage of providing a scheme for handling acoustic feedback that can adapt relatively fast to changing acoustic environments. The invention may e.g. be used in listening devices, such as hearing aids, head sets or active ear plugs, wherein customized feedback compensation is an issue.
摘要翻译：本发明涉及一种用于根据用户需要处理输出声音到输出声音的听力仪器。本发明还涉及一种操作听力仪器和使用听力仪器的方法。本发明的目的是提供一种用于处理听力仪器中的声反馈的替代方案。解决的问题在于，用于将输入声音转换为电输入信号的输入传感器和用于将经处理的电输出信号转换为输出声音的输出换能器，在输入换能器和输出换能器之间限定正向通路，反馈消除系统，用于估计从输出变换器到输入换能器的声反馈的影响，反馈消除系统包括可变预估滤波器和存储器，其中多个预定反馈信道脉冲响应对应于多个声环境，经历的实质性反馈被存储，并且其中听力仪器包括基于当前声学环境的监测单元，适于在存储的脉冲响应中选择当前最合适的可变预估滤波器的脉冲响应。这具有提供用于处理可以相对快速地适应变化的声学环境适应的声反馈的方案的优点。本发明可以例如用于听力装置，例如助听器，头戴式耳塞或有源耳塞，其中定制的反馈补偿是一个问题。

6. 发明申请

US20120087509A1 METHOD OF DETERMINING PARAMETERS IN AN ADAPTIVE AUDIO PROCESSING ALGORITHM AND AN AUDIO PROCESSING SYSTEM 有权
标题翻译：在自适应音频处理算法和音频处理系统中确定参数的方法
公开(公告)号：US20120087509A1
公开(公告)日：2012-04-12
申请号：US13267624
申请日：2011-10-06
申请人： Thomas Bo ELMEDYB , Jesper JENSEN , Meng GUO
发明人： Thomas Bo ELMEDYB , Jesper JENSEN , Meng GUO
IPC分类号： G10K11/16
CPC分类号： H04R3/02 , H04R25/453 , H04R2430/20
摘要： A method and an audio processing system determine a system parameter, e.g. step size, in an adaptive algorithm, e.g. an adaptive feedback cancellation algorithm so as to provide an alternative scheme for feedback estimation in a multi-microphone audio processing system. A feedback part of the system's open loop transfer function is estimated and separated in a transient part and a steady state part, which can be used to control the adaptation rate of the adaptive feedback cancellation algorithm by adjusting the system parameter, e.g. step size parameter, of the algorithm when desired system properties, such as a steady state value or a convergence rate of the feedback, are given/desired. The method can be used for different adaptation algorithms such as LMS, NLMS, RLS, etc. in hearing aids, headsets, handsfree telephone systems, teleconferencing systems, public address systems, etc.
摘要翻译：方法和音频处理系统确定系统参数，例如，步长，在自适应算法中，例如。自适应反馈消除算法，以便在多麦克风音频处理系统中提供用于反馈估计的替代方案。系统的开环传递函数的反馈部分在瞬态部分和稳态部分中进行估计和分离，稳态部分可用于通过调整系统参数来控制自适应反馈消除算法的自适应速率。给定/期望所需的系统特性（如稳定状态值或反馈的收敛速度）时的算法的步长参数。该方法可用于助听器，耳机，免提电话系统，电话会议系统，公共广播系统等不同适应算法，如LMS，NLMS，RLS等。

7. 发明申请

US20100020981A1 SPECTRAL CONTENT MODIFICATION FOR ROBUST FEEDBACK CHANNEL ESTIMATION 有权
公开(公告)号：US20100020981A1
公开(公告)日：2010-01-28
申请号：US12506424
申请日：2009-07-21
申请人： Thomas Bo ELMEDYB , Jesper JENSEN
发明人： Thomas Bo ELMEDYB , Jesper JENSEN
IPC分类号： G10K11/16 , A61F11/06
CPC分类号： H04R25/353 , H04R25/453
摘要： The invention relates to a listening device for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of reducing acoustic feedback in a listening device and to the use of a listening device. The object of the present invention is to provide an alternative scheme for feedback estimation in a listening device. The problem is solved in that the listening device comprises an input transducer for converting an input sound to an electric input signal, and an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and comprising a signal processing unit adapted for processing an SPU-input signal originating from the electric input signal in a time-frequency representation comprising successive time frames each comprising a frequency spectrum of the signal in the time frame in question, the signal processing unit defining an input side and an output side of the forward path and comprising a spectral content modification unit adapted for modifying values of the signal of one or more regions of the frequency spectrum of a given time frame to provide that the modified values are less correlated to the corresponding time-frequency regions of the input signal than the unmodified output signal thereby providing an improved processed output signal, and a feedback loop from the output side to the input side comprising a feedback path estimation unit for estimating the effect of acoustic feedback from the output transducer to the input transducer, wherein the feedback path estimation unit is adapted to use the improved processed output signal in the estimation. This has the advantage of providing a better accuracy vs. tracking speed trade-off of the feedback path estimate. The invention may e.g. be used for listening devices prone to acoustic feedback, e.g. hearing aids, headsets or active earplugs.

8. 发明授权

US08958587B2 Signal dereverberation using environment information 有权
标题翻译：信号失真使用环境信息
公开(公告)号：US08958587B2
公开(公告)日：2015-02-17
申请号：US13089053
申请日：2011-04-18
申请人： Jesper Jensen
发明人： Jesper Jensen
IPC分类号： H04R25/00 , G01H7/00 , G10L15/20 , H04S7/00 , H03G3/00 , G10L21/0208
CPC分类号： G01H7/00 , G10L15/20 , G10L2021/02082 , H04R25/407 , H04R25/554 , H04R2225/39 , H04R2225/41 , H04R2225/43 , H04R2225/55 , H04S7/305
摘要： An audio processing system includes an audio processing device and an external device, the audio processing device having a signal processing unit adapted for running an algorithm for processing an input signal representing an acoustic signal from the environment of the user and providing a processed output signal. The external device includes in an information signal a measure, characteristic of the reverberation of the room or location where the audio processing device is located, and the audio processing device is adapted to extract the measure of the reverberation from the information signal and use the measure as an input to an algorithm that includes a directional algorithm for providing a directional characteristic of the input signal.
摘要翻译：音频处理系统包括音频处理设备和外部设备，音频处理设备具有信号处理单元，该信号处理单元适于运行用于处理来自用户环境的表示声学信号的输入信号的算法，并提供经处理的输出信号。外部设备在信息信号中包括测量，音频处理设备所在的房间或位置的混响的特征，并且音频处理设备适于从信息信号中提取混响的度量并使用该测量作为包括用于提供输入信号的方向特性的定向算法的算法的输入。

9. 发明申请

US20110311075A1 SPECTRAL BAND SUBSTITUTION TO AVOID HOWLS AND SUB-OSCILLATION 有权
标题翻译：光谱带取代避免HOWLS和SUB-OSCILLATION
公开(公告)号：US20110311075A1
公开(公告)日：2011-12-22
申请号：US13146849
申请日：2009-02-06
申请人： Thomas Bo Elmedyb , Jesper Jensen
发明人： Thomas Bo Elmedyb , Jesper Jensen
IPC分类号： H04B15/00
CPC分类号： H04R25/453 , H04R3/02 , H04R25/353 , H04R2430/03
摘要： A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion. Based on an input from the LG-estimator unit, the SBS unit is adapted for substituting spectral content in a receiver band of the input signal with spectral content from a donor band in such a way that spectral content of the donor band is copied and possibly scaled with a scaling function and inserted in the receiver band instead of its original spectral content, wherein the receiver band is a plus-band and the donor band is a minus-band.
摘要翻译：用于处理输出声音到输出声音的收听装置包括用于将输入声音转换为电输入信号的输入换能器，用于将经处理的电输出信号转换为输出声音的输出换能器，输入传感器和输出变换器，并且包括用于处理多个频带中的输入信号的信号处理单元和用于从一个频带到另一个频带进行频谱带替换并提供经SBS处理的输出信号的SBS单元，以及LG 估计单元，用于估计每个频带中的环路增益，从而根据具有根据负准则的估计环路增益的正准则和负频带来识别具有估计环路增益的加频带。基于来自LG估计器单元的输入，SBS单元适于用来自施主频带的频谱内容来替代输入信号的接收机频带中的频谱内容，使得复制供体频带的频谱内容，并且可能用缩放函数缩放并插入接收机频带而不是其原始频谱内容，其中接收机频带是正频带，供体频带是负频带。

10. 发明申请

US20100020979A1 ADAPTIVE LONG-TERM PREDICTION FILTER FOR ADAPTIVE WHITENING 有权
标题翻译：适应性长期自适应长期预测滤波器
公开(公告)号：US20100020979A1
公开(公告)日：2010-01-28
申请号：US12506983
申请日：2009-07-21
申请人： Thomas Bo ELMEDYB , Jesper Jensen
发明人： Thomas Bo ELMEDYB , Jesper Jensen
IPC分类号： H04R29/00 , H04R25/00
CPC分类号： H04R25/453 , H04R5/033
摘要： The invention relates to a hearing instrument for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of estimating acoustic feedback in a hearing instrument. The object of the present invention is to reduce the impact of tonal components in the target input signal on the quality of the estimate of acoustic feedback. The problem is solved in that hearing instrument comprises an input transducer for converting an input sound to an electric input signal and an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and comprising a signal processing unit defining an input side and an output side of the forward path, a feedback loop from the output side to the input side comprising a feedback cancellation system for estimating the effect of acoustic feedback from the output transducer to the input transducer, the feedback cancellation system comprising a feedback path estimation unit receiving first and second estimation input signals from the input and output side of the forward path, respectively, wherein the first and second estimation input signal paths comprise first and second long term prediction filters P(z) each having an input and an output, the feedback cancellation system being adapted to provide that the variable parameters of at least one of the long term prediction filters are estimated based on the input signal to the filter in question. Embodiments of the invention have the advantage of leading to better feedback cancellation, even for tonal input signals. The invention may e.g. be used for the hearing instruments or headsets.
摘要翻译：本发明涉及一种用于根据用户需要处理输出声音到输出声音的听力仪器。本发明还涉及一种在听力仪器中估计声反馈的方法。本发明的目的是减少目标输入信号中的音调分量对声反馈估计质量的影响。解决的问题在于，听力仪器包括用于将输入声音转换为电输入信号的输入换能器和用于将经处理的电输出信号转换为输出声音的输出换能器，在输入换能器和输出端之间限定正向通路换能器并且包括限定正向路径的输入侧和输出侧的信号处理单元，从输出侧到输入侧的反馈回路包括用于估计从输出换能器到输入端的声反馈的影响的反馈消除系统所述反馈消除系统包括反馈路径估计单元，其分别从所述正向路径的输入和输出侧接收第一和第二估计输入信号，其中所述第一和第二估计输入信号路径包括第一和第二长期预测滤波器P （z），每个具有输入和输出，反馈消除系统bei ng适于提供基于所讨论的滤波器的输入信号来估计长期预测滤波器中的至少一个的可变参数。本发明的实施例具有导致更好的反馈消除的优点，即使对于音调输入信号也是如此。本发明可以例如用于听力仪器或耳机。

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