Efficient dereverberation in networked audio systems
    15.
    发明授权
    Efficient dereverberation in networked audio systems 有权
    网络音频系统中的高效混响

    公开(公告)号:US09390723B1

    公开(公告)日:2016-07-12

    申请号:US14568033

    申请日:2014-12-11

    CPC classification number: G10K11/175 G10L21/0208 G10L21/0232 G10L2021/02082

    Abstract: Features are disclosed for performing efficient dereverberation of speech signals captured with single- and multi-channel sensors in networked audio systems. Such features could be used in applications requiring automatic recognition of speech captured with sensors. Dereverberation is performed in the sub-band domain, and hence provides improved dereverberation performance in terms of signal quality, algorithmic delay, computational efficiency, and speed of convergence.

    Abstract translation: 公开了用于对网络音频系统中的单通道和多通道传感器捕获的语音信号进行有效的去混响的特征。 这些特征可以用于需要用传感器捕获的语音自动识别的应用中。 在子带域中执行混频,从而在信号质量,算法延迟,计算效率和收敛速度方面提供改进的去混响性能。

    Parallel noise suppression
    16.
    发明授权

    公开(公告)号:US11792570B1

    公开(公告)日:2023-10-17

    申请号:US17470035

    申请日:2021-09-09

    Abstract: Techniques for improving microphone noise suppression are provided. As wind noise may disproportionately impact a subset of microphones, a method for processing audio data using two adaptive reference algorithm (ARA) paths in parallel is provided. For example, first ARA processing performs noise cancellation using all microphones, while second ARA processing performs noise cancellation using only a portion of the microphones. As the first ARA processing and the second ARA processing are performed in parallel, beam merging can be performed using beams from the first ARA, the second ARA, and/or a combination of each. In addition, beam merging can be performed using beam sections instead of individual beams to further improve performance and reduce attenuation to speech.

    Echo cancellation by acoustic playback estimation

    公开(公告)号:US10937418B1

    公开(公告)日:2021-03-02

    申请号:US16240294

    申请日:2019-01-04

    Abstract: A system configured to improve echo cancellation for nonlinear systems. The system generate reference audio data by isolating portions of microphone audio data that correspond to playback audio data. For example, the system may determine a correlation between the playback audio data and the microphone audio data in individual time-frequency bands in a frequency domain. In some examples, the system may substitute microphone audio data associated with output audio for the playback audio data. The system may generate the reference audio data based on portions of the microphone audio data that have a strong correlation with the playback audio data. The system may generate the reference audio data by selecting these portions of the microphone audio data or by performing beamforming. This results in precise time alignment between the reference audio data and the microphone audio data, improving performance of the echo cancellation.

    Multichannel noise cancellation using deep neural network masking

    公开(公告)号:US10522167B1

    公开(公告)日:2019-12-31

    申请号:US15895313

    申请日:2018-02-13

    Abstract: A system configured to improve beamforming by using deep neural networks (DNNs). The system can use one trained DNN to focus on a first person speaking an utterance (e.g., target user) and one or more trained DNNs to focus on noise source(s) (e.g., wireless loudspeaker(s), a second person speaking, other localized sources of noise, or the like). The DNNs may generate time-frequency mask data that indicates individual frequency bands that correspond to the particular source detected by the DNN. Using this mask data, a beamformer can generate beamformed audio data that is specific to a source of noise. The system may perform noise cancellation to isolate first beamformed audio data associated with the target user by removing second beamformed audio data associated with noise source(s).

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