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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：US08190426B2
公开(公告)日：2012-05-29
申请号：US11948137
申请日：2007-11-30
申请人： Mohamed Krini , Gerhard Uwe Schmidt
发明人： Mohamed Krini , Gerhard Uwe Schmidt
IPC分类号： G10L19/02
CPC分类号： G10L19/0204 , G10L21/02 , G10L25/18 , G10L25/27
摘要： An audio enhancement refines a short-time spectrum. The refinement may reduce overlap between audio sub-bands. The sub-bands are transformed into sub-band short-time spectra. A portion of the spectra are time-delayed. The sub-band short-time spectrum and the time-delayed portion are filtered to obtain a refined sub-band short-time spectrum. The refined spectrum improves audio processing.
摘要翻译： 音频增强优化了短时间频谱。 细化可以减少音频子带之间的重叠。 子带被转换成子带短时光谱。 光谱的一部分是时间延迟的。 子带短时频谱和时间延迟部分被滤波以获得精确的子带短时频谱。 精致的频谱改进了音频处理。
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公开(公告)号：US07554465B2
公开(公告)日：2009-06-30
申请号：US11855575
申请日：2007-09-14
IPC分类号： H03M7/40
CPC分类号： H03H17/0642 , H03H17/0685
摘要： A sampling rate conversion system reduces the signal processing burdens carried by cellular phones, headsets, and other electronic devices. Because the system consumes fewer resources to convert between signal sampling rates, the system may significantly reduce processing time and resource requirements in the device. As a result, the device may instead devote resources to performing other useful tasks, such as interacting with the user through a graphical user interface and performing selected processing tasks.
摘要翻译： 采样率转换系统减少了蜂窝电话，耳机和其他电子设备携带的信号处理负担。 由于系统消耗较少的资源以在信号采样率之间进行转换，所以系统可能会显着减少设备中的处理时间和资源需求。 因此，设备可以替代地投入资源来执行其他有用的任务，例如通过图形用户界面与用户交互并执行所选择的处理任务。
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公开(公告)号：US20080151984A1
公开(公告)日：2008-06-26
申请号：US11855575
申请日：2007-09-14
IPC分类号： H03K5/159
CPC分类号： H03H17/0642 , H03H17/0685
摘要： A sampling rate conversion system reduces the signal processing burdens carried by cellular phones, headsets, and other electronic devices. Because the system consumes fewer resources to convert between signal sampling rates, the system may significantly reduce processing time and resource requirements in the device. As a result, the device may instead devote resources to performing other useful tasks, such as interacting with the user through a graphical user interface and performing selected processing tasks.
摘要翻译： 采样率转换系统减少了蜂窝电话，耳机和其他电子设备携带的信号处理负担。 由于系统消耗较少的资源以在信号采样率之间进行转换，所以系统可能会显着减少设备中的处理时间和资源需求。 因此，设备可以替代地投入资源来执行其他有用的任务，例如通过图形用户界面与用户交互并执行所选择的处理任务。
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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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6.发明申请SYSTEM FOR PROCESSING AN ACOUSTIC INPUT SIGNAL TO PROVIDE AN OUTPUT SIGNAL WITH REDUCED NOISE 有权用于处理声音输入信号以减少噪声的输出信号的系统公开(公告)号：US20080304679A1
公开(公告)日：2008-12-11
申请号：US12118205
申请日：2008-05-09
IPC分类号： H04B15/00
CPC分类号： G10L21/0208 , G10L21/0232 , G10L2021/02168
摘要： An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.
摘要翻译： 一种装置处理声输入信号以提供具有降低的噪声的输出信号。 该装置基于频率依赖加权函数对输入信号进行加权。 频率相关阈值函数从下面界定加权函数。
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7.发明授权System for processing an acoustic input signal to provide an output signal with reduced noise 有权用于处理声输入信号以提供具有降低的噪声的输出信号的系统公开(公告)号：US08199928B2
公开(公告)日：2012-06-12
申请号：US12118205
申请日：2008-05-09
IPC分类号： H04B15/00
CPC分类号： G10L21/0208 , G10L21/0232 , G10L2021/02168
摘要： An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.
摘要翻译： 一种装置处理声输入信号以提供具有降低的噪声的输出信号。 该装置基于频率依赖加权函数对输入信号进行加权。 频率相关阈值函数从下面界定加权函数。
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公开(公告)号：US20120109647A1
公开(公告)日：2012-05-03
申请号：US13273890
申请日：2011-10-14
申请人： Gerhard Uwe Schmidt , Mohamed Krini
发明人： Gerhard Uwe Schmidt , Mohamed Krini
IPC分类号： G10L15/20
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：US20090216526A1
公开(公告)日：2009-08-27
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
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：US20090119096A1
公开(公告)日：2009-05-07
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
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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