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1. 发明授权

US08218780B2 Methods and systems for blind dereverberation 有权
标题翻译：盲目混响的方法和系统
公开(公告)号：US08218780B2
公开(公告)日：2012-07-10
申请号：US12484686
申请日：2009-06-15
申请人： Thomas Anthony Baran , Bowon Lee , Ronald W. Schafer , Majid Fozunbal
发明人： Thomas Anthony Baran , Bowon Lee , Ronald W. Schafer , Majid Fozunbal
IPC分类号： H04B3/20 , H03G3/00
CPC分类号： H04M9/082
摘要： Various embodiments of the present invention are directed to methods for dereverberation of audio generated in a room. In one aspect, a method for dereverberating reverberant digital signals comprises transforming a reverberant digital signal from the time domain into Fourier domain signals using a computing device, each Fourier domain signal corresponding to a subband. For each subband of the Fourier domain signal, the method computes autoregressive model coefficients of the reverberation with the current and previous magnitudes of the Fourier digital signal, and inverse filters the magnitude of the Fourier domain signal using the computing device, based on the autoregressive model coefficients and previous magnitudes of the Fourier digital signal. The method includes inverse transforming the Fourier domain signals with filtered magnitudes into an approximate dereverberated digital signal.
摘要翻译：本发明的各种实施例涉及用于在室内产生的音频的混响的方法。一方面，一种用于去混响混响数字信号的方法包括使用计算装置将混响数字信号从时域变换成傅立叶域信号，每个傅立叶域信号对应于子带。对于傅立叶域信号的每个子带，该方法利用傅里叶数字信号的当前和先前幅度来计算混响的自回归模型系数，并且使用计算装置基于自回归模型对傅立叶域信号的幅度进行滤波傅里叶数字信号的系数和先前幅度。该方法包括将具有滤波幅度的傅立叶域信号逆变换为近似的非反相数字信号。

2. 发明申请

US20100316228A1 METHODS AND SYSTEMS FOR BLIND DEREVERBERATION 有权
标题翻译： BLIND DEREVERBERATION的方法和系统
公开(公告)号：US20100316228A1
公开(公告)日：2010-12-16
申请号：US12484686
申请日：2009-06-15
申请人： Thomas Anthony Baran , Bowon Lee , Ronald W. Schafer , Majid Fozunbal
发明人： Thomas Anthony Baran , Bowon Lee , Ronald W. Schafer , Majid Fozunbal
IPC分类号： H04B3/20
CPC分类号： H04M9/082
摘要： Various embodiments of the present invention are directed to methods for dereverberation of audio generated in a room. In one aspect, a method for dereverberating reverberant digital signals comprises transforming a reverberant digital signal from the time domain into Fourier domain signals using a computing device, each Fourier domain signal corresponding to a subband. For each subband of the Fourier domain signal, the method computes autoregressive model coefficients of the reverberation with the current and previous magnitudes of the Fourier digital signal, and inverse filters the magnitude of the Fourier domain signal using the computing device, based on the autoregressive model coefficients and previous magnitudes of the Fourier digital signal. The method includes inverse transforming the Fourier domain signals with filtered magnitudes into an approximate dereverberated digital signal.
摘要翻译：本发明的各种实施例涉及用于在室内产生的音频的混响的方法。一方面，一种用于去混响混响数字信号的方法包括使用计算装置将混响数字信号从时域变换成傅立叶域信号，每个傅立叶域信号对应于子带。对于傅立叶域信号的每个子带，该方法利用傅里叶数字信号的当前和先前幅度来计算混响的自回归模型系数，并且使用计算装置基于自回归模型对傅立叶域信号的幅度进行滤波傅里叶数字信号的系数和先前幅度。该方法包括将具有滤波幅度的傅立叶域信号逆变换为近似的非反相数字信号。

3. 发明授权

US08483398B2 Methods and systems for reducing acoustic echoes in multichannel communication systems by reducing the dimensionality of the space of impulse responses 有权
标题翻译：通过降低脉冲响应空间的维度来减少多通道通信系统中的声学回波的方法和系统
公开(公告)号：US08483398B2
公开(公告)日：2013-07-09
申请号：US12387351
申请日：2009-04-30
申请人： Majid Fozunbal , Ronald W. Schafer
发明人： Majid Fozunbal , Ronald W. Schafer
IPC分类号： H04B3/20 , H04B15/00 , H04M9/08
CPC分类号： H04M9/082
摘要： Various embodiments of the present invention are directed to adaptive methods for reducing acoustic echoes in multichannel audio communication systems. Acoustic echo cancellation methods determine approximate impulse responses characterizing each echo path between loudspeakers and microphones within a room and improve performance based on previously determined impulse responses. In particular, the methods adapt to changes in the room by inferring approximate impulse responses that lie within a model of an impulse response space. Over time the method improves performance by evolving the model into a more accurate space from which to select subsequent approximate impulse responses.
摘要翻译：本发明的各种实施例涉及用于减少多声道音频通信系统中的声学回波的自适应方法。声回波消除方法确定表征房间内扬声器和麦克风之间的每个回波路径的近似脉冲响应，并且基于先前确定的脉冲响应提高性能。特别地，该方法通过推断位于脉冲响应空间的模型内的近似脉冲响应来适应房间的变化。随着时间的推移，该方法通过将模型演变为更准确的空间来提高性能，从中选择随后的近似脉冲响应。

4. 发明申请

US20100278351A1 Methods and systems for reducing acoustic echoes in multichannel communication systems by reducing the dimensionality of the space of impulse resopnses 有权
标题翻译：通过降低脉冲振荡空间的维度来减少多通道通信系统中的声学回波的方法和系统
公开(公告)号：US20100278351A1
公开(公告)日：2010-11-04
申请号：US12387351
申请日：2009-04-30
申请人： Majid Fozunbal , Ronald W. Schafer
发明人： Majid Fozunbal , Ronald W. Schafer
IPC分类号： H04B3/20
CPC分类号： H04M9/082
摘要： Various embodiments of the present invention are directed to adaptive methods for reducing acoustic echoes in multichannel audio communication systems. Acoustic echo cancellation methods determine approximate impulse responses characterizing each echo path between loudspeakers and microphones within a room and improve performance based on previously determined impulse responses. In particular, the methods adapt to changes in the room by inferring approximate impulse responses that lie within a model of an impulse response space. Over time the method improves performance by evolving the model into a more accurate space from which to select subsequent approximate impulse responses.
摘要翻译：本发明的各种实施例涉及用于减少多声道音频通信系统中的声学回波的自适应方法。声回波消除方法确定表征房间内扬声器和麦克风之间的每个回波路径的近似脉冲响应，并且基于先前确定的脉冲响应提高性能。特别地，该方法通过推断位于脉冲响应空间的模型内的近似脉冲响应来适应房间的变化。随着时间的推移，该方法通过将模型演变为更准确的空间来提高性能，从中选择随后的近似脉冲响应。

5. 发明授权

US08320574B2 Methods and systems for reducing acoustic echoes in communication systems 有权
标题翻译：用于减少通信系统中声学回声的方法和系统
公开(公告)号：US08320574B2
公开(公告)日：2012-11-27
申请号：US11786481
申请日：2007-04-12
申请人： Majid Fozunbal
发明人： Majid Fozunbal
IPC分类号： H04B3/20 , H04B15/00 , H04R3/00 , H04R27/00 , H03F99/00
CPC分类号： H04M9/082
摘要： Various embodiments of the present invention are directed to methods and systems that reduce acoustic echoes in audio signals in accordance with changing conditions at first and second locations that are linked together in a communication system. In particular, one embodiment of the present invention relates to a method for determining an approximate impulse-response vector for canceling an acoustic echo resulting from an audio signal transmitted from the first location to the second location. This method includes forming a trust region within a search space based on computing a recursive specification vector defining the trust region. The method also includes computing a recursive shadow-impulse-response vector that lies substantially within the trust region, and computing the approximate impulse-response vector based on the recursive shadow-impulse-response vector and the recursive specification vector.
摘要翻译：本发明的各种实施例涉及根据通信系统中链接在一起的第一和第二位置处的变化条件来减少音频信号中的声学回声的方法和系统。特别地，本发明的一个实施例涉及一种用于确定用于消除由从第一位置传输到第二位置的音频信号产生的声学回声的近似脉冲响应向量的方法。该方法包括：基于计算定义信任区域的递归规范向量，在搜索空间内形成信任区域。该方法还包括计算基本上在信任区域内的递归阴影脉冲响应向量，以及基于递归阴影 - 脉冲 - 响应向量和递归规范向量计算近似脉冲响应向量。

6. 发明授权

US08515094B2 Distributed signal processing systems and methods 有权
标题翻译：分布式信号处理系统和方法
公开(公告)号：US08515094B2
公开(公告)日：2013-08-20
申请号：US12902907
申请日：2010-10-12
申请人： Amir Said , Ton Kalker , Majid Fozunbal
发明人： Amir Said , Ton Kalker , Majid Fozunbal
IPC分类号： H04R3/00
CPC分类号： H04R3/005 , G01S3/808 , H04R1/406 , H04R2201/403 , H04R2420/07
摘要： Systems and methods for parallel and distributed processing of audio signals produced by a microphone array are described. In one aspect, a distributed signal processing system includes an array of microphones and an array of processors. Each processor is connected to one of the microphones and is connected to at least two other processors, enabling communication between adjacent connected processors. The system also includes a computing device connected to each of the processors. Each microphone detects a sound and generates an audio signal, and each processor is configured to receive and process the audio signal sent from a connected microphone and audio signals sent from at least one of the adjacent processors to produce a data stream that is sent to the computing device.
摘要翻译：描述了由麦克风阵列产生的音频信号并行和分布式处理的系统和方法。在一个方面，分布式信号处理系统包括麦克风阵列和处理器阵列。每个处理器连接到一个麦克风，并且连接到至少两个其他处理器，实现相邻连接的处理器之间的通信。该系统还包括连接到每个处理器的计算设备。每个麦克风检测声音并产生音频信号，并且每个处理器被配置为接收和处理从连接的麦克风发送的音频信号和从至少一个相邻处理器发送的音频信号，以产生发送到计算设备。

7. 发明授权

US08208649B2 Methods and systems for robust approximations of impulse responses in multichannel audio-communication systems 有权
标题翻译：用于多通道音频通信系统中脉冲响应鲁棒逼近的方法和系统
公开(公告)号：US08208649B2
公开(公告)日：2012-06-26
申请号：US12387075
申请日：2009-04-28
申请人： Majid Fozunbal , Ton Kalker
发明人： Majid Fozunbal , Ton Kalker
IPC分类号： H04B15/00 , H04R3/00 , H04R27/00 , H03F99/00
CPC分类号： H04M9/085
摘要： Proper estimation of impulse responses is an important and challenging aspect of multichannel echo control. Various embodiments of the present invention are directed to real-time, adaptive acoustic echo cancellation methods in multichannel audio-communication systems. In particular, embodiments of the present invention use a collection of a room's past impulse responses to determine an optimal lower dimensional impulse response space as the underlying search subspace for approximate impulse responses. As a result, embodiments of the present invention mitigate inherent instability of multichannel audio-communication systems and provide stable and accurate echo removal without distorting audio signals.
摘要翻译：脉冲响应的适当估计是多通道回波控制的一个重要且具有挑战性的方面。本发明的各种实施例涉及多声道音频通信系统中的实时自适应声学回声消除方法。具体地，本发明的实施例使用房间过去脉冲响应的集合来确定最佳较低维度脉冲响应空间作为用于近似脉冲响应的底层搜索子空间。结果，本发明的实施例减轻了多声道音频通信系统的固有不稳定性，并且提供了稳定且准确的回波消除而不会使音频信号失真。

8. 发明授权

US08204249B2 Methods and systems for reducing acoustic echoes in multichannel audio-communication systems 有权
标题翻译：用于减少多声道音频通信系统中的声学回波的方法和系统
公开(公告)号：US08204249B2
公开(公告)日：2012-06-19
申请号：US11799266
申请日：2007-04-30
申请人： Majid Fozunbal
发明人： Majid Fozunbal
IPC分类号： H04B15/00 , H04R3/00 , H04R27/00
CPC分类号： H04M9/082
摘要： Various embodiments of the present invention are directed to adaptive real-time, acoustic echo cancellation methods and systems. One method embodiment of the present invention is directed to reducing acoustic echoes in microphone-digital signals transmitted from a first location to a second location. The first location includes a plurality of loudspeakers and microphones, each microphone produces one of the microphone-digital signals including sounds produced at the first location and acoustic echoes produced by the loudspeakers. The method includes determining approximate impulse responses, each of which corresponds to an echo path between the microphones and the loudspeakers. The method includes determining a plurality of approximate acoustic echoes, each approximate acoustic echo corresponds to convolving a digital signal played by one of the loudspeakers with a number of the approximate impulse responses. The acoustic echo in at least one of the microphone-digital signals is reduced based on the corresponding approximate acoustic echo.
摘要翻译：本发明的各种实施例涉及自适应实时声学回声消除方法和系统。本发明的一个方法实施例旨在减少从第一位置传输到第二位置的麦克风数字信号中的声学回声。第一位置包括多个扬声器和麦克风，每个麦克风产生麦克风数字信号之一，包括在第一位置产生的声音和由扬声器产生的声学回声。该方法包括确定近似脉冲响应，其中每个脉冲响应对应于麦克风和扬声器之间的回波路径。该方法包括确定多个近似声学回波，每个近似声学回声对应于使用多个近似脉冲响应的扬声器之一播放的数字信号进行卷积。基于对应的近似声学回声，至少一个麦克风数字信号中的声学回声被减小。

9. 发明授权

US09538299B2 Acoustic echo cancellation (AEC) with conferencing environment templates (CETs) 有权
标题翻译：具有会议环境模板（CET）的声学回声消除（AEC）
公开(公告)号：US09538299B2
公开(公告)日：2017-01-03
申请号：US12606930
申请日：2009-10-27
申请人： Byron A. Alcorn , Majid Fozunbal
发明人： Byron A. Alcorn , Majid Fozunbal
IPC分类号： H04R27/00 , H04M9/08
CPC分类号： H04R27/00 , G06F15/16 , H04M9/082
摘要： In at least some embodiments, a computer system includes a processor and a network interface coupled to the processor. The computer system also includes a system memory coupled to the processor, the system memory storing a communication application having a conferencing user interface. The conferencing user interface, when executed, enables a user to set up a conferencing session by selecting one of a plurality of conferencing environment templates (CETs). Each CET is matched to one of a plurality of different sets of acoustic echo cancellation (AEC) parameters to be applied during the conferencing session.
摘要翻译：在至少一些实施例中，计算机系统包括耦合到处理器的处理器和网络接口。计算机系统还包括耦合到处理器的系统存储器，系统存储器存储具有会议用户界面的通信应用。会议用户界面在被执行时使用户能够通过选择多个会议环境模板（CET）中的一个来建立会议会话。每个CET与在会议会话期间应用的多个不同组的声学回声消除（AEC）参数中的一个匹配。

10. 发明申请

US20120087512A1 DISTRIBUTED SIGNAL PROCESSING SYSTEMS AND METHODS 有权
标题翻译：分布式信号处理系统和方法
公开(公告)号：US20120087512A1
公开(公告)日：2012-04-12
申请号：US12902907
申请日：2010-10-12
申请人： Amir Said , Ton Kalker , Majid Fozunbal
发明人： Amir Said , Ton Kalker , Majid Fozunbal
IPC分类号： H04R3/00
CPC分类号： H04R3/005 , G01S3/808 , H04R1/406 , H04R2201/403 , H04R2420/07
摘要： Systems and methods for parallel and distributed processing of audio signals produced by a microphone array are described. In one aspect, a distributed signal processing system includes an array of microphones and an array of processors. Each processor is connected to one of the microphones and is connected to at least two other processors, enabling communication between adjacent connected processors. The system also includes a computing device connected to each of the processors. Each microphone detects a sound and generates an audio signal, and each processor is configured to receive and process the audio signal sent from a connected microphone and audio signals sent from at least one of the adjacent processors to produce a data stream that is sent to the computing device.
摘要翻译：描述了由麦克风阵列产生的音频信号并行和分布式处理的系统和方法。在一个方面，分布式信号处理系统包括麦克风阵列和处理器阵列。每个处理器连接到一个麦克风，并且连接到至少两个其他处理器，实现相邻连接的处理器之间的通信。该系统还包括连接到每个处理器的计算设备。每个麦克风检测声音并产生音频信号，并且每个处理器被配置为接收和处理从连接的麦克风发送的音频信号和从至少一个相邻处理器发送的音频信号，以产生发送到计算设备。

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