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    • 3. 发明申请
    • System and Method for Multi-Channel Noise Suppression
    • 多通道噪声抑制系统与方法
    • US20120123773A1
    • 2012-05-17
    • US13295889
    • 2011-11-14
    • Huaiyu ZENGJes ThyssenNelson SollenbergerJuin-Hwey ChenXianxian Zhang
    • Huaiyu ZENGJes ThyssenNelson SollenbergerJuin-Hwey ChenXianxian Zhang
    • G10L21/02
    • G10L21/0208G10L21/0272G10L2021/02165H04R1/245H04R2410/07
    • 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.
    • 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。
    • 10. 发明授权
    • Dynamic time scale modification for reduced bit rate audio coding
    • 用于降低比特率音频编码的动态时间尺度修改
    • US08670990B2
    • 2014-03-11
    • US12847120
    • 2010-07-30
    • Juin-Hwey ChenHong-Goo KangRobert W. ZopfJes Thyssen
    • Juin-Hwey ChenHong-Goo KangRobert W. ZopfJes Thyssen
    • G10L21/04G10L11/06
    • G10L19/22G10L19/08
    • 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.
    • 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。