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公开(公告)号：US20130208905A1
公开(公告)日：2013-08-15
申请号：US13591667
申请日：2012-08-22
申请人： Mohamed Krini , Gerhard Schmidt , Bernd Iser , Arthur Wolf
发明人： Mohamed Krini , Gerhard Schmidt , Bernd Iser , Arthur Wolf
IPC分类号： G10L21/0208
CPC分类号： G10L21/0208 , G10K11/002 , G10L19/0204 , G10L21/02 , G10L2021/02082
摘要： Embodiments of the present invention exploit redundancy of succeeding FFT spectra and use this redundancy for computing interpolated temporal supporting points. An analysis filter bank converts overlapped sequences of an audio (ex. loudspeaker) signal from a time domain to a frequency domain to obtain a time series of short-time loudspeaker spectra. An interpolator temporally interpolates this time series. The interpolation is fed to an echo canceller, which computes an estimated echo spectrum. A microphone analysis filter bank converts overlapped sequences of an audio microphone signal from the time domain to the frequency domain to obtain a time series of short-time microphone spectra. The estimated echo spectrum is subtracted from the microphone spectrum. Further signal enhancement (filtration) may be applied. A synthesis filter bank converts the filtered microphone spectra to the time domain to generate an echo compensated audio microphone signal. Computational complexity of signal processing systems can, therefore, be reduced.
摘要翻译： 本发明的实施例利用后续FFT频谱的冗余并使用该冗余来计算内插的时间支持点。 分析滤波器组将来自时域的音频（例如扬声器）信号的重叠序列转换为频域以获得短时间扬声器频谱的时间序列。 内插器对时间序列进行时间插值。 内插被馈送到回波消除器，该消除器计算估计的回波频谱。 麦克风分析滤波器组将音频麦克风信号的重叠序列从时域转换到频域，以获得短时间麦克风频谱的时间序列。 从麦克风频谱中减去估计回波频谱。 可以应用进一步的信号增强（过滤）。 合成滤波器组将滤波的麦克风谱转换为时域以产生回波补偿的音频麦克风信号。 因此，可以减少信号处理系统的计算复杂度。
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公开(公告)号：US09076455B2
公开(公告)日：2015-07-07
申请号：US13591667
申请日：2012-08-22
申请人： Mohamed Krini , Gerhard Schmidt , Bernd Iser , Arthur Wolf
发明人： Mohamed Krini , Gerhard Schmidt , Bernd Iser , Arthur Wolf
IPC分类号： G10L21/0208 , G10K11/00 , G10L21/02 , G10L19/02
CPC分类号： G10L21/0208 , G10K11/002 , G10L19/0204 , G10L21/02 , G10L2021/02082
摘要： Embodiments of the present invention exploit redundancy of succeeding FFT spectra and use this redundancy for computing interpolated temporal supporting points. An analysis filter bank converts overlapped sequences of an audio (ex. loudspeaker) signal from a time domain to a frequency domain to obtain a time series of short-time loudspeaker spectra. An interpolator temporally interpolates this time series. The interpolation is fed to an echo canceller, which computes an estimated echo spectrum. A microphone analysis filter bank converts overlapped sequences of an audio microphone signal from the time domain to the frequency domain to obtain a time series of short-time microphone spectra. The estimated echo spectrum is subtracted from the microphone spectrum. Further signal enhancement (filtration) may be applied. A synthesis filter bank converts the filtered microphone spectra to the time domain to generate an echo compensated audio microphone signal. Computational complexity of signal processing systems can, therefore, be reduced.
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公开(公告)号：US20100036659A1
公开(公告)日：2010-02-11
申请号：US12537749
申请日：2009-08-07
申请人： Tim Haulick , Mohamed Krini , Shreyas Paranjpe , Gerhard Schmidt
发明人： Tim Haulick , Mohamed Krini , Shreyas Paranjpe , Gerhard Schmidt
CPC分类号： G10L19/012 , G10L21/0208
摘要： The present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.
摘要翻译： 本发明涉及一种用于信号处理的方法，包括以下步骤：提供一组原型频谱包络，提供一组参考噪声原型，其中参考噪声原型从所提供的一组原型的至少一个子集获得 通过至少一个麦克风检测语音话语以获得麦克风信号，基于所提供的参考噪声原型处理麦克风信号以进行噪声降低，以获得增强信号，并基于所提供的原型频谱包络对增强信号进行编码， 获得编码的增强信号。
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公开(公告)号：US09026435B2
公开(公告)日：2015-05-05
申请号：US12772562
申请日：2010-05-03
申请人： Mohamed Krini , Gerhard Schmidt
发明人： Mohamed Krini , Gerhard Schmidt
IPC分类号： G10L19/09 , G10L25/90 , G10L21/0216
CPC分类号： G10L25/90 , G10L2021/02168
摘要： The invention provides a method for estimating a fundamental frequency of a speech signal comprising the steps of receiving a signal spectrum of the speech signal, filtering the signal spectrum to obtain a refined signal spectrum, determining a cross-power spectral density using the refined signal spectrum and the signal spectrum, transforming the cross-power spectral density into the time domain to obtain a cross-correlation function, and estimating the fundamental frequency of the speech signal based on the cross-correlation function.
摘要翻译： 本发明提供了一种用于估计语音信号的基频的方法，包括以下步骤：接收语音信号的信号频谱，滤波信号频谱以获得精细的信号频谱，使用精细的信号频谱确定交叉功率频谱密度 和信号频谱，将交叉功率谱密度变换成时域以获得互相关函数，并且基于互相关函数估计语音信号的基频。
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公开(公告)号：US08666736B2
公开(公告)日：2014-03-04
申请号：US12537749
申请日：2009-08-07
申请人： Tim Haulick , Mohamed Krini , Shreyas Paranjpe , Gerhard Schmidt
发明人： Tim Haulick , Mohamed Krini , Shreyas Paranjpe , Gerhard Schmidt
IPC分类号： G10L21/00
CPC分类号： G10L19/012 , G10L21/0208
摘要： The present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.
摘要翻译： 本发明涉及一种用于信号处理的方法，包括以下步骤：提供一组原型频谱包络，提供一组参考噪声原型，其中参考噪声原型从所提供的一组原型的至少一个子集获得 通过至少一个麦克风检测语音话语以获得麦克风信号，基于所提供的参考噪声原型处理麦克风信号以进行噪声降低，以获得增强信号，并且基于所提供的原型频谱包络对增强信号进行编码， 获得编码的增强信号。
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公开(公告)号：US08849656B2
公开(公告)日：2014-09-30
申请号：US13273890
申请日：2011-10-14
申请人： Gerhard Schmidt , Mohamed Krini
发明人： Gerhard Schmidt , Mohamed Krini
IPC分类号： G10L21/02 , G10L21/0208 , H04R3/00 , G10L21/0216 , G10L21/0264
CPC分类号： H04R3/005 , G10L21/0208 , G10L21/0264 , G10L2021/02165 , H04R2410/05 , H04R2410/07 , H04R2499/11 , H04R2499/13
摘要： A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.
摘要翻译： 通过从位于距离干扰源的第一距离的第一麦克风检测扬声器的话语来增强语音。 第二麦克风可以在不同的位置检测说话者的话语。 监视装置可以估计第一麦克风信号的功率电平。 合成器可以通过处理第二麦克风信号来合成第一麦克风信号的一部分。 当功率水平低于预定水平时，合成可能发生。
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公开(公告)号：US20100286981A1
公开(公告)日：2010-11-11
申请号：US12772562
申请日：2010-05-03
申请人： Mohamed Krini , Gerhard Schmidt
发明人： Mohamed Krini , Gerhard Schmidt
IPC分类号： G10L21/00
CPC分类号： G10L25/90 , G10L2021/02168
摘要： The invention provides a method for estimating a fundamental frequency of a speech signal comprising the steps of receiving a signal spectrum of the speech signal, filtering the signal spectrum to obtain a refined signal spectrum, determining a cross-power spectral density using the refined signal spectrum and the signal spectrum, transforming the cross-power spectral density into the time domain to obtain a cross-correlation function, and estimating the fundamental frequency of the speech signal based on the cross-correlation function.
摘要翻译： 本发明提供了一种用于估计语音信号的基频的方法，包括以下步骤：接收语音信号的信号频谱，滤波信号频谱以获得精细的信号频谱，使用精细的信号频谱确定交叉功率频谱密度 和信号频谱，将交叉功率谱密度变换成时域以获得互相关函数，并且基于互相关函数估计语音信号的基频。
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公开(公告)号：US09805738B2
公开(公告)日：2017-10-31
申请号：US14423543
申请日：2012-09-04
申请人： Mohamed Krini , Ingo Schalk-Schupp , Markus Buck
发明人： Mohamed Krini , Ingo Schalk-Schupp , Markus Buck
CPC分类号： G10L25/18 , G10L19/06 , G10L21/02 , G10L21/0232 , G10L2019/0016
摘要： An arrangement is described for speech signal processing. An input microphone signal is received that includes a speech signal component and a noise component. The microphone signal is transformed into a frequency domain set of short-term spectra signals. Then speech formant components within the spectra signals are estimated based on detecting regions of high energy density in the spectra signals. One or more dynamically adjusted gain factors are applied to the spectra signals to enhance the speech formant components.
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公开(公告)号：US08050914B2
公开(公告)日：2011-11-01
申请号：US12269605
申请日：2008-11-12
申请人： Gerhard Uwe Schmidt , Mohamed Krini
发明人： Gerhard Uwe Schmidt , Mohamed Krini
IPC分类号： G10L21/02
CPC分类号： H04R3/005 , G10L21/0208 , G10L21/0264 , G10L2021/02165 , H04R2410/05 , H04R2410/07 , H04R2499/11 , H04R2499/13
摘要： A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.
摘要翻译： 通过从位于距离干扰源的第一距离的第一麦克风检测扬声器的话语来增强语音。 第二麦克风可以在不同的位置检测说话者的话语。 监视装置可以估计第一麦克风信号的功率电平。 合成器可以通过处理第二麦克风信号来合成第一麦克风信号的一部分。 当功率水平低于预定水平时，合成可能发生。
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公开(公告)号：US08706483B2
公开(公告)日：2014-04-22
申请号：US12254488
申请日：2008-10-20
申请人： Franz Gerl , Tobias Herbig , Mohamed Krini , Gerhard Uwe Schmidt
发明人： Franz Gerl , Tobias Herbig , Mohamed Krini , Gerhard Uwe Schmidt
IPC分类号： G10L21/00
CPC分类号： H04R3/005 , G10L21/0208 , G10L21/0264 , G10L2021/02165 , H04R2410/05 , H04R2410/07 , H04R2499/11 , H04R2499/13
摘要： A system enhances the quality of a digital speech signal that may include noise. The system identifies vocal expressions that correspond to the digital speech signal. A signal-to-noise ratio of the digital speech signal is measured before a portion of the digital speech signal is synthesized. The selected portion of the digital speech signal may have a signal-to-noise ratio below a predetermined level and the synthesis of the digital speech signal may be based on speaker identification.
摘要翻译： 系统提高可能包括噪声的数字语音信号的质量。 该系统识别对应于数字语音信号的声乐表达。 在数字语音信号的一部分被合成之前测量数字语音信号的信噪比。 数字语音信号的所选部分可以具有低于预定电平的信噪比，并且数字语音信号的合成可以基于说话者识别。
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