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1. 发明申请

US20130231925A1 Monaural Noise Suppression Based on Computational Auditory Scene Analysis 有权
标题翻译：基于计算听觉场景分析的单声道噪声抑制
公开(公告)号：US20130231925A1
公开(公告)日：2013-09-05
申请号：US13859186
申请日：2013-04-09
申请人： Carlos Avendano , Jean Laroche , Michael M. Goodwin , Ludger Solbach
发明人： Carlos Avendano , Jean Laroche , Michael M. Goodwin , Ludger Solbach
IPC分类号： G10L21/0208
CPC分类号： G10L21/0208 , G10L21/0272
摘要： The present technology provides a robust noise suppression system that may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. A time-domain acoustic signal may be received and be transformed to frequency-domain sub-band signals. Features, such as pitch, may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals. An acoustic signal may be reconstructed from the noise-reduced sub-band signals.
摘要翻译：本技术提供了一种强大的噪声抑制系统，其可以同时降低声信号中的噪声和回波分量，同时限制语音失真的水平。可以接收时域声信号并将其变换到频域子带信号。可以在子带信号内识别和跟踪诸如音调的特征。可以至少部分地基于跟踪的音调源的概率分析来估计初始语音和噪声模型。语音和噪声模型可以从初始语音和噪声模型中解决，并且可以对子带信号执行噪声降低。可以从噪声降低的子带信号重构声信号。

2. 发明授权

US08204253B1 Self calibration of audio device 有权
标题翻译：音频设备的自校准
公开(公告)号：US08204253B1
公开(公告)日：2012-06-19
申请号：US12286909
申请日：2008-10-02
申请人： Ludger Solbach
发明人： Ludger Solbach
IPC分类号： H04B15/00 , G10L21/02
CPC分类号： H04R3/005 , H04M9/082 , H04R2410/01
摘要： An audio device performs self calibration with respect to an audio source location when processing an audio signal frame determined likely to be dominated by the audio source. One or more conditions for sub-bands within the audio frame are evaluated to help identify whether the frame is dominated by the audio source. If the conditions meet a threshold value for a number of sub-bands within the frame, the audio signal may be identified as one dominated by the desired audio source and an audio source location coefficient may be adapted. Additionally, when the audio source location coefficient falls below a threshold value, (e.g., suggesting that one of two or more microphones is blocked), noise suppression is reduced or eliminated for the frame or frame sub-bands to prevent suppression of a desired audio source component along with the noise component.
摘要翻译：当处理可能由音频源主导的音频信号帧时，音频设备相对于音频源位置执行自校准。评估音频帧内的子带的一个或多个条件以帮助识别帧是否被音频源支配。如果条件满足帧内的多个子带的阈值，则音频信号可以被识别为由所需音频源支配的音频信号，并且可以调整音频源位置系数。此外，当音频源位置系数低于阈值时（例如，暗示两个或多个麦克风中的一个被阻止），对于帧或帧子带，减少或消除噪声抑制以防止抑制所需的音频源分量以及噪声分量。

3. 发明授权

US08194882B2 System and method for providing single microphone noise suppression fallback 有权
标题翻译：提供单麦克风噪声抑制后备的系统和方法
公开(公告)号：US08194882B2
公开(公告)日：2012-06-05
申请号：US12072931
申请日：2008-02-29
申请人： Mark Every , Carlos Avendano , Ludger Solbach , Carlo Murgia
发明人： Mark Every , Carlos Avendano , Ludger Solbach , Carlo Murgia
IPC分类号： H04B15/00
CPC分类号： G10L21/0208 , G10L2021/02165
摘要： Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.
摘要翻译：提供了提供单麦克风噪声抑制后备的系统和方法。在示例性实施例中，接收主和次声信号。可以基于主声信号来生成单个麦克风噪声估计，而可以基于主声信号和次声信号来生成双麦克风噪声估计。然后确定基于单个和双重麦克风噪声估计的组合噪声估计。使用组合噪声估计，可以生成增益掩模并将其应用于主声信号以产生噪声抑制信号。随后，可以输出噪声抑制信号。

4. 发明授权

US08949120B1 Adaptive noise cancelation 有权
标题翻译：自适应噪声消除
公开(公告)号：US08949120B1
公开(公告)日：2015-02-03
申请号：US12422917
申请日：2009-04-13
申请人： Mark Every , Ludger Solbach , Carlo Murgia , Ye Jiang
发明人： Mark Every , Ludger Solbach , Carlo Murgia , Ye Jiang
IPC分类号： G10L21/00 , G10L15/00 , G10K11/16 , H04B15/00 , G10L21/0208 , G10L21/0316 , G10L21/02
CPC分类号： G10L21/0208 , G10L21/02 , G10L21/0316 , G10L21/034 , G10L2021/02166
摘要： Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.
摘要翻译：提出了用于控制噪声抵消适应性的系统和方法。一个或多个音频信号由一个或多个相应的麦克风接收。一个或多个信号可以被分解成频率子带。对所述一个或多个音频信号执行与所识别的适应约束一致的噪声消除。然后可以从频率子带重构一个或多个音频信号，并通过输出设备输出。

5. 发明授权

US08774423B1 System and method for controlling adaptivity of signal modification using a phantom coefficient 有权
标题翻译：使用幻影系数控制信号修改的适应性的系统和方法
公开(公告)号：US08774423B1
公开(公告)日：2014-07-08
申请号：US12286995
申请日：2008-10-02
申请人： Ludger Solbach
发明人： Ludger Solbach
IPC分类号： H04B15/00
CPC分类号： H04R3/005 , H04R2410/01
摘要： Systems and methods for controlling adaptivity of signal modification, such as noise suppression, using a phantom coefficient are provided. The process for controlling adaptivity comprises receiving a signal. Determinations may be made of whether an adaptation coefficient satisfies an adaptation constraint and of whether the phantom coefficient satisfies the adaptation constraint. The phantom coefficient may be updated, for example, toward a current observation. The adaptation coefficient may be updated, for example, toward the phantom coefficient, based on whether the phantom coefficient satisfies an adaptation constraint of the signal. A modified signal may be generated by applying the adaptation coefficient to the signal based on whether the adaptation coefficient satisfies the adaptation constraint. Accordingly, the modified signal may be outputted.
摘要翻译：提供了用于控制信号修改的适应性的系统和方法，例如使用幻影系数的噪声抑制。用于控制自适应性的过程包括接收信号。可以确定自适应系数是否满足适应约束以及幻影系数是否满足适应约束。例如，幻影系数可以被更新为当前的观察。基于幻影系数是否满足信号的适应约束，可以将适配系数例如更新为虚拟系数。可以通过基于自适应系数是否满足自适应约束将适配系数应用于信号来生成修改的信号。因此，可以输出修改的信号。

6. 发明授权

US08150065B2 System and method for processing an audio signal 有权
标题翻译：用于处理音频信号的系统和方法
公开(公告)号：US08150065B2
公开(公告)日：2012-04-03
申请号：US11441675
申请日：2006-05-25
申请人： Ludger Solbach , Lloyd Watts
发明人： Ludger Solbach , Lloyd Watts
IPC分类号： H03G5/00
CPC分类号： G10L19/0204 , H04R25/505 , H04R2430/03
摘要： Systems and methods for audio signal processing are provided. In exemplary embodiments, a filter cascade of complex-valued filters are used to decompose an input audio signal into a plurality of frequency components or sub-band signals. These sub-band signals may be processed for phase alignment, amplitude compensation, and time delay prior to summation of real portions of the sub-band signals to generate a reconstructed audio signal.
摘要翻译：提供音频信号处理的系统和方法。在示例性实施例中，复值滤波器的滤波器级联用于将输入音频信号分解为多个频率分量或子带信号。在对子带信号的实部进行求和之前，可以对这些子带信号进行相位对准，幅度补偿和时间延迟处理，以产生重构的音频信号。

7. 发明申请

US20100094643A1 Systems and methods for reconstructing decomposed audio signals 有权
标题翻译：用于重建分解的音频信号的系统和方法
公开(公告)号：US20100094643A1
公开(公告)日：2010-04-15
申请号：US12319107
申请日：2008-12-31
申请人： Carlos Avendano , Ludger Solbach
发明人： Carlos Avendano , Ludger Solbach
IPC分类号： G10L19/00
CPC分类号： G10L19/0204 , G10L25/18
摘要： Systems and methods for reconstructing decomposed audio signals are presented. In exemplary embodiments, a decomposed audio signal is received. The decomposed audio signal may include a plurality of frequency sub-band signals having successively shifted group delays as a function of frequency from a filter bank. The plurality of frequency sub-band signals may then be grouped into two or more groups. A delay function may be applied to at least one of the two or more groups. Subsequently, the groups may be combined to reconstruct the audio signal, which may be outputted accordingly.
摘要翻译：提出了重建分解音频信号的系统和方法。在示例性实施例中，接收分解的音频信号。分解的音频信号可以包括具有作为来自滤波器组的频率的函数的连续移位的组延迟的多个频率子带信号。然后可以将多个频率子带信号分组为两个或更多个组。延迟功能可以应用于两个或更多个组中的至少一个。随后，可以组合组以重建可相应地输出的音频信号。

8. 发明申请

US20090220107A1 System and method for providing single microphone noise suppression fallback 有权
标题翻译：提供单麦克风噪声抑制后备的系统和方法
公开(公告)号：US20090220107A1
公开(公告)日：2009-09-03
申请号：US12072931
申请日：2008-02-29
申请人： Mark Every , Carlos Avendano , Ludger Solbach , Carlo Murgia
发明人： Mark Every , Carlos Avendano , Ludger Solbach , Carlo Murgia
IPC分类号： H04B15/00
CPC分类号： G10L21/0208 , G10L2021/02165
摘要： Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.
摘要翻译：提供了提供单麦克风噪声抑制后备的系统和方法。在示例性实施例中，接收主和次声信号。可以基于主声信号来产生单个麦克风噪声估计，而可以基于主声信号和次声信号来生成双麦克风噪声估计。然后确定基于单个和双重麦克风噪声估计的组合噪声估计。使用组合噪声估计，可以生成增益掩模并将其应用于主声信号以产生噪声抑制信号。随后，可以输出噪声抑制信号。

9. 发明申请

US20070276656A1 System and method for processing an audio signal 有权
标题翻译：用于处理音频信号的系统和方法
公开(公告)号：US20070276656A1
公开(公告)日：2007-11-29
申请号：US11441675
申请日：2006-05-25
申请人： Ludger Solbach , Lloyd Watts
发明人： Ludger Solbach , Lloyd Watts
IPC分类号： G10L19/00
CPC分类号： G10L19/0204 , H04R25/505 , H04R2430/03
摘要： Systems and methods for audio signal processing are provided. In exemplary embodiments, a filter cascade of complex-valued filters are used to decompose an input audio signal into a plurality of frequency components or sub-band signals. These sub-band signals may be processed for phase alignment, amplitude compensation, and time delay prior to summation of real portions of the sub-band signals to generate a reconstructed audio signal.
摘要翻译：提供音频信号处理的系统和方法。在示例性实施例中，复值滤波器的滤波器级联用于将输入音频信号分解为多个频率分量或子带信号。在对子带信号的实部进行求和之前，可以对这些子带信号进行相位对准，幅度补偿和时间延迟处理，以产生重构的音频信号。

10. 发明授权

US09378754B1 Adaptive spatial classifier for multi-microphone systems 有权
标题翻译：多麦克风系统的自适应空间分类器
公开(公告)号：US09378754B1
公开(公告)日：2016-06-28
申请号：US12841061
申请日：2010-07-21
申请人： Mark Every , Ludger Solbach
发明人： Mark Every , Ludger Solbach
IPC分类号： G10L21/00 , G10L21/0272 , H04B3/20 , G10L21/0208 , G10L21/0216
CPC分类号： G10L21/0272 , G10L21/0208 , G10L25/18 , G10L25/81 , G10L2021/02082 , G10L2021/02166 , H04B3/20
摘要： A robust noise suppression system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to cochlea domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.
摘要翻译：鲁棒的噪声抑制系统可以同时降低声信号中的噪声和回波分量，同时限制语音失真的水平。系统可以以紧密通话，手持式或其他配置从两个或更多个麦克风接收声信号。接收到的声信号被变换为耳蜗域子带信号，并且可以从子带信号中减去回波和噪声分量。识别声学子带信号中的特征并用于产生乘法掩模。将乘法掩码应用于噪声减去的子带信号，并且在时域中重构子带信号。

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