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    Audio source parameterization
    11.
    发明授权
    Audio source parameterization 有权

    公开(公告)号：US11152014B2

    公开(公告)日：2021-10-19

    申请号：US16090739

    申请日：2017-04-05

    Applicant: Dolby Laboratories Licensing Corporation

    Inventor： Jun Wang

    IPC: G10L21/0308 ,  G10L21/0272 ,  G10L19/08 ,  H04S3/00

    Abstract: The present document describes a method (600) for estimating source parameters of audio sources (101) from mix audio signals (102), with. The mix audio signals (102) comprise a plurality of frames. The mix audio signals (102) are representable as a mix audio matrix in a frequency domain and the audio sources (101) are representable as a source matrix in the frequency domain. The method (600) comprises updating (601) an un-mixing matrix (221) which is configured to provide an estimate of the source matrix from the mix audio matrix, based on a mixing matrix (225) which is configured to provide an estimate of the mix audio matrix from the source matrix. Furthermore, the method (600) comprises updating (602) the mixing matrix (225) based on the un-mixing matrix (221) and based on the mix audio signals (102). In addition, the method (600) comprises iterating (603) the updating steps (601, 602) until an overall convergence criteria is met.

    Source separation for reverberant environment
    13.
    发明授权
    Source separation for reverberant environment 有权

    公开(公告)号：US10667069B2

    公开(公告)日：2020-05-26

    申请号：US16323763

    申请日：2017-08-28

    Applicant: Dolby Laboratories Licensing Corporation

    Inventor： Jun Wang

    IPC: H04S3/00 ,  H04R3/00 ,  G10L21/028 ,  G10L25/21 ,  H04R1/40 ,  G10L25/18 ,  H03H21/00

    Abstract: Embodiments of source separation for reverberant environment are disclosed. According to a method, first microphone signals for each individual one of at least one source are captured respectively by at least two microphones for a period during which only the individual one produces sounds. Mixing parameters for modeling acoustic paths between the at least one source and the at least two microphones are learned by a processor based on the first microphone signals. Second microphone signals are captured respectively by the at least two microphones for a period during which all of the at least one source produce sounds. The reconstruction model is estimated by the processor based on the mixing parameters and second microphone signals. The processor performs the source separation by applying the reconstruction model.

    Decomposing audio signals
    14.
    发明授权
    Decomposing audio signals 有权

    公开(公告)号：US10650836B2

    公开(公告)日：2020-05-12

    申请号：US16577467

    申请日：2019-09-20

    Applicant: DOLBY LABORATORIES LICENSING CORPORATION

    Inventor： Jun Wang ,  Lie Lu

    IPC: G10L19/02 ,  G10L25/21 ,  G10L19/008

    Abstract: Example embodiments disclosed herein relate to signal processing. A method for decomposing a plurality of audio signals from at least two different channels is disclosed. The method comprises obtaining a set of components that are weakly correlated, the set of components generated based on the plurality of audio signals. The method comprises extracting a feature from the set of components, and determining a set of gains associated with the set of components at least in part based on the extracted feature, each of the gains indicating a proportion of a diffuse part in the associated component. The method further comprises decomposing the plurality of audio signals by applying the set of gains to the set of components. Corresponding system and computer program product are also disclosed.

    Adaptive panner of audio objects
    15.
    发明授权
    Adaptive panner of audio objects 有权

    公开(公告)号：US10405120B2

    公开(公告)日：2019-09-03

    申请号：US15647121

    申请日：2017-07-11

    Applicant: Dolby Laboratories Licensing Corporation ,  Dolby International AB

    Inventor： Jun Wang ,  Giulio Cengarle ,  Juan Felix Torres ,  Daniel Arteaga

    IPC: H04R5/02 ,  H04S3/00 ,  H04S7/00 ,  H04R5/04

    Abstract: An audio object including audio content and object metadata is received. The object metadata indicates an object spatial position of the audio object to be rendered by audio speakers in a playback environment. Based on the object spatial position and source spatial positions of the audio speakers, initial gain values for the audio speakers are determined. The initial gain values can be used to select a set of audio speakers from among the audio speakers. Based on the object spatial position and a set of source spatial positions at which the set of audio speakers are respectively located in the playback environment, a set of non-negative optimized gain values for the set of audio speakers is determined. The audio object at the object spatial position is rendered with the set of optimized gain values for the set of audio speakers.

    Volume leveler controller and controlling method
    19.
    发明授权
    Volume leveler controller and controlling method 有权
    卷矫平控制器及控制方法

    公开(公告)号：US09548713B2

    公开(公告)日：2017-01-17

    申请号：US14777271

    申请日：2014-03-17

    Applicant: DOLBY LABORATORIES LICENSING CORPORATION

    Inventor： Jun Wang ,  Lie Lu ,  Alan Seefeldt

    IPC: H03G3/00 ,  H03G3/32 ,  H03G3/30 ,  H03G7/00 ,  H03G5/16

    CPC classification number: H03G7/002 ,  G10L21/0364 ,  G10L25/30 ,  G10L25/51 ,  H03G3/3089 ,  H03G3/32 ,  H03G5/165 ,  H03G7/007 ,  H04M7/006 ,  H04M2203/305

    Abstract: Volume leveler controller and controlling method are disclosed. In one embodiment, A volume leveler controller includes an audio content classifier for identifying the content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified. The adjusting unit may configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.

    Abstract translation: 公开了卷积矫直机控制器和控制方法。 在一个实施例中，音量调平器控制器包括用于实时地识别音频信号的内容类型的音频内容分类器; 以及调整单元，用于基于所识别的内容类型以连续的方式调整音量调节器。 调整单元可以被配置为使音量调平器的动态增益与音频信号的信息内容类型正相关，并且将音量调平器的动态增益与音频信号的干扰内容类型负相关。

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