公开(公告)号：US20120123772A1

公开(公告)日：2012-05-17

申请号：US13295818

申请日：2011-11-14

申请人： Jes Thyssen ,  Huaiyu Zeng ,  Juin-Hwey Chen ,  Nelson Sollenberger ,  Xianxian Zhang

发明人： Jes Thyssen ,  Huaiyu Zeng ,  Juin-Hwey Chen ,  Nelson Sollenberger ,  Xianxian Zhang

IPC分类号： G10L21/02

CPC分类号： G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

摘要： Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

摘要翻译： 描述了多通道噪声抑制系统和方法，其省略了传统的延迟和总和固定波束形成器，其包括主话音麦克风和至少一个噪声参考麦克风，其中所需语音位于设备的近场。 多声道噪声抑制系统和方法使用阻塞矩阵（BM）去除由噪声参考麦克风接收的输入语音信号中的期望语音以获得“更干净”的背景噪声分量。 然后，使用自适应噪声消除器（ANC）来基于“更干净的”背景噪声分量来消除由主话音麦克风接收的输入语音信号中的背景噪声，以实现噪声抑制。 由BM和ANC实现的滤波器是使用需要在噪声抑制系统中计算复杂频域信号的时变统计的闭式解决方案导出的。

公开(公告)号：US08965757B2

公开(公告)日：2015-02-24

申请号：US13295818

申请日：2011-11-14

申请人： Jes Thyssen ,  Huaiyu Zeng ,  Juin-Hwey Chen ,  Nelson Sollenberger ,  Xianxian Zhang

发明人： Jes Thyssen ,  Huaiyu Zeng ,  Juin-Hwey Chen ,  Nelson Sollenberger ,  Xianxian Zhang

IPC分类号： G10L21/02 ,  G10L21/0208 ,  G10L21/0272 ,  H04R1/24 ,  G10L21/0216

CPC分类号： G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

摘要： Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

摘要翻译： 描述了多通道噪声抑制系统和方法，其省略了传统的延迟和总和固定波束形成器，其包括主话音麦克风和至少一个噪声参考麦克风，其中所需语音位于设备的近场。 多声道噪声抑制系统和方法使用阻塞矩阵（BM）去除由噪声参考麦克风接收的输入语音信号中的期望语音以获得“更干净”的背景噪声分量。 然后，使用自适应噪声消除器（ANC）来基于“更干净的”背景噪声分量来消除由主话音麦克风接收的输入语音信号中的背景噪声，以实现噪声抑制。 由BM和ANC实现的滤波器是使用需要在噪声抑制系统中计算复杂频域信号的时变统计的闭式解决方案导出的。

公开(公告)号：US20120123773A1

公开(公告)日：2012-05-17

申请号：US13295889

申请日：2011-11-14

申请人： Huaiyu ZENG ,  Jes Thyssen ,  Nelson Sollenberger ,  Juin-Hwey Chen ,  Xianxian Zhang

发明人： Huaiyu ZENG ,  Jes Thyssen ,  Nelson Sollenberger ,  Juin-Hwey Chen ,  Xianxian Zhang

IPC分类号： G10L21/02

CPC分类号： G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

摘要： Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

摘要翻译： 这里描述的是多通道噪声抑制系统和方法，其被配置为使用至少两个空间分离的麦克风来检测和抑制风和背景噪声：至少一个主话音麦克风和至少一个噪声参考麦克风。 在至少一个示例中，多声道噪声抑制系统和方法被配置为首先检测和抑制由主话音麦克风拾取的输入语音信号中的风噪声，并且潜在地由噪声拾取的输入语音信号 参考麦克风 在风噪声检测和抑制之后，多声道噪声抑制系统和方法被配置为在两个阶段中执行进一步的噪声抑制：包括阻塞矩阵和自适应噪声消除器的第一线性处理级，随后是第二非线性 处理阶段。

公开(公告)号：US08977545B2

公开(公告)日：2015-03-10

申请号：US13295889

申请日：2011-11-14

申请人： Huaiyu Zeng ,  Jes Thyssen ,  Nelson Sollenberger ,  Juin-Hwey Chen ,  Xianxian Zhang

发明人： Huaiyu Zeng ,  Jes Thyssen ,  Nelson Sollenberger ,  Juin-Hwey Chen ,  Xianxian Zhang

IPC分类号： G10L21/02 ,  G10L21/0208 ,  G10L21/0272 ,  H04R1/24 ,  G10L21/0216

CPC分类号： G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

摘要： Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

摘要翻译： 这里描述的是多通道噪声抑制系统和方法，其被配置为使用至少两个空间分离的麦克风来检测和抑制风和背景噪声：至少一个主话音麦克风和至少一个噪声参考麦克风。 在至少一个示例中，多声道噪声抑制系统和方法被配置为首先检测和抑制由主话音麦克风拾取的输入语音信号中的风噪声，并且潜在地由噪声拾取的输入语音信号 参考麦克风 在风噪声检测和抑制之后，多声道噪声抑制系统和方法被配置为在两个阶段中执行进一步的噪声抑制：包括阻塞矩阵和自适应噪声消除器的第一线性处理级，随后是第二非线性 处理阶段。

公开(公告)号：US09330675B2

公开(公告)日：2016-05-03

申请号：US13250355

申请日：2011-09-30

申请人： Xianxian Zhang ,  Juin-Hwey Chen ,  Huaiyu Zeng ,  Jes Thyssen

发明人： Xianxian Zhang ,  Juin-Hwey Chen ,  Huaiyu Zeng ,  Jes Thyssen

IPC分类号： A61F11/06 ,  G10L21/0208 ,  G10L21/0272 ,  H04R1/24 ,  G10L21/0216

CPC分类号： G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

摘要： Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this tact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

摘要翻译： 不同于声音的压力波，无处不在，由风引起的空气湍流通常是一个相当地方的事件。 因此，在使用两个或多个空间分离的麦克风拾取声音信号（例如语音）的系统中，由麦克风之一拾取的风噪声通常不会被拾取（或至少不相同的程度），由 另一个麦克风。 描述利用这种技巧和其他技术来有效地检测和抑制使用空间分离的多个麦克风的风噪声的方法和装置的实施例。

公开(公告)号：US20120121100A1

公开(公告)日：2012-05-17

申请号：US13250355

申请日：2011-09-30

申请人： Xianxian ZHANG ,  Juin-Hwey Chen ,  Huaiyu Zeng ,  Jes Thyssen

发明人： Xianxian ZHANG ,  Juin-Hwey Chen ,  Huaiyu Zeng ,  Jes Thyssen

IPC分类号： G10K11/16

CPC分类号： G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

摘要： Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this tact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

摘要翻译： 不同于声音的压力波，无处不在，由风引起的空气湍流通常是一个相当地方的事件。 因此，在使用两个或多个空间分离的麦克风拾取声音信号（例如语音）的系统中，由麦克风之一拾取的风噪声通常不会被拾取（或至少不相同的程度），由 另一个麦克风。 描述利用这种技巧和其他技术来有效地检测和抑制使用空间分离的多个麦克风的风噪声的方法和装置的实施例。

公开(公告)号：US08924204B2

公开(公告)日：2014-12-30

申请号：US13250291

申请日：2011-09-30

申请人： Juin-Hwey Chen ,  Jes Thyssen ,  Xianxian Zhang ,  Huaiyu Zeng

发明人： Juin-Hwey Chen ,  Jes Thyssen ,  Xianxian Zhang ,  Huaiyu Zeng

IPC分类号： G10L21/00 ,  G10L21/02 ,  G10L15/00 ,  G10L15/20 ,  G10L13/00 ,  G10L25/93 ,  G10L19/00 ,  G10L19/12 ,  G10L19/02 ,  G10L25/90 ,  G10L21/0208 ,  H04R1/24 ,  G10L13/06 ,  H04R3/00 ,  H04R29/00 ,  H03G3/00 ,  H04R5/00 ,  H04R5/02 ,  H03G3/20 ,  H04B15/00 ,  H04B17/00 ,  H04W24/00 ,  G10L21/0216

CPC分类号： G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

摘要： Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

摘要翻译： 不同于声音的压力波，无处不在，由风引起的空气湍流通常是一个相当地方的事件。 因此，在使用两个或多个空间分离的麦克风拾取声音信号（例如语音）的系统中，由麦克风之一拾取的风噪声通常不会被拾取（或至少不相同的程度），由 另一个麦克风。 描述了利用这一事实和其他方法和装置来有效地检测和抑制使用空间分离的多个麦克风的风噪声的方法和装置的实施例。

公开(公告)号：US09336785B2

公开(公告)日：2016-05-10

申请号：US12464498

申请日：2009-05-12

申请人： Jes Thyssen ,  Wilfrid LeBlanc ,  Juin-Hwey Chen

发明人： Jes Thyssen ,  Wilfrid LeBlanc ,  Juin-Hwey Chen

IPC分类号： G10L19/14 ,  G10L19/012 ,  G10L21/0208 ,  G10L21/0232

CPC分类号： G10L21/0208 ,  G10L19/012 ,  G10L21/0232

摘要： A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

摘要翻译： 描述了一种语音清晰度增强（SIE）系统和方法，当音频设备位于具有大声背景噪声的环境中时，提高了由音频设备重放的语音信号的可懂度。 在一个实施例中，音频设备包括近端电话终端，并且语音信号包括通过通信网络从远端电话终端接收的用于在近端电话终端回放的语音信号。

公开(公告)号：US09130643B2

公开(公告)日：2015-09-08

申请号：US13362244

申请日：2012-01-31

申请人： Juin-Hwey Chen ,  Thomas F. Baker ,  Evan S. McCarthy ,  Jes Thyssen ,  Walter J. Wihardja ,  David C. Garrett

发明人： Juin-Hwey Chen ,  Thomas F. Baker ,  Evan S. McCarthy ,  Jes Thyssen ,  Walter J. Wihardja ,  David C. Garrett

IPC分类号： H04B1/10 ,  H03D1/04 ,  H04B1/16 ,  H04B1/7183 ,  H03G3/34 ,  H04B1/04

CPC分类号： H04B1/1661 ,  H03G3/345 ,  H04B1/04 ,  H04B1/7183

摘要： Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, quadrature L−R demodulation is applied to a composite baseband signal output by an FM demodulator to obtain an L−R noise signal. A channel quality measure is calculated based on the L−R noise signal and is used to control whether a pop suppression technique is applied to an L+R signal obtained from the composite baseband signal to detect and remove noise pulses therefrom. The channel quality measure and the L−R noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on an L−R signal obtained from the composite baseband signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L−R and L+R signal with replacement waveforms generated via waveform extrapolation.

摘要翻译： 描述了用于增强FM接收机的音频质量的系统和方法。 在本文描述的实施例中，正交L-R解调被应用于由FM解调器输出的复合基带信号以获得L-R噪声信号。 基于L-R噪声信号计算信道质量度量，并且用于控制是否将弹出抑制技术应用于从复合基带信号获得的L + R信号，以从其检测和去除噪声脉冲。 信道质量测量和L-R噪声信号也被用于在从复合基带信号和L + R信号获得的L-R信号上对频域执行单信道噪声抑制。 信道质量测量还用于控制快速衰落补偿过程的应用，其中替换了通过波形外推产生的替换波形的L-R和L + R信号的噪声段。

公开(公告)号：US08670990B2

公开(公告)日：2014-03-11

申请号：US12847120

申请日：2010-07-30

申请人： Juin-Hwey Chen ,  Hong-Goo Kang ,  Robert W. Zopf ,  Jes Thyssen

发明人： Juin-Hwey Chen ,  Hong-Goo Kang ,  Robert W. Zopf ,  Jes Thyssen

IPC分类号： G10L21/04 ,  G10L11/06

CPC分类号： G10L19/22 ,  G10L19/08

摘要： Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.

摘要翻译： 描述了利用动态时间尺度修正（TSM）来实现降低的比特率音频编码的系统和方法。 根据实施例，在由编码器对TSM压缩进行编码之前，将不同级别的TSM压缩选择性地应用于输入语音信号的段。 编码的TSM压缩段在解码器处被接收，解码器对这些段进行解码，然后基于从编码器接收的信息向每个TSM解压缩应用适当级别的TSM解压缩。 通过在编码之前选择性地对输入语音信号的段应用不同级别的TSM压缩，减少与编码器/解码器相关联的编码比特率。 此外，通过选择考虑到该信号的某些局部特性的输入语音信号的每个段的TSM压缩级别，提供这样的比特率降低，而不会将不可接受的失真电平引入到由解码器产生的输出语音信号中。