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    POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT
    32.
    发明申请
    POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT 审中-公开

    公开(公告)号：US20170162212A1

    公开(公告)日：2017-06-08

    申请号：US15433437

    申请日：2017-02-15

    Applicant: Dolby Laboratories Licensing Corporation

    Inventor： Xuejing SUN ,  Glenn N. DICKINS

    IPC: G10L21/0364 ,  G10L21/0224 ,  G10L21/034 ,  H03G3/30 ,  G10L25/78

    CPC classification number: G10L21/0364 ,  G10K11/16 ,  G10L21/0224 ,  G10L21/0316 ,  G10L21/034 ,  G10L25/78 ,  H03G3/301 ,  H03G3/32

    Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

    JITTER BUFFER LEVEL ESTIMATION
    33.
    发明申请
    JITTER BUFFER LEVEL ESTIMATION 审中-公开

    公开(公告)号：US20170026298A1

    公开(公告)日：2017-01-26

    申请号：US15125564

    申请日：2015-04-08

    Applicant: DOLBY LABORATORIES LICENSING CORPORATION

    Inventor： JiaQuan HUO ,  Xuejing SUN ,  Kai LI

    IPC: H04L12/841 ,  H04L12/835 ,  H04L12/26

    CPC classification number: H04L47/283 ,  H04J3/0632 ,  H04L43/087 ,  H04L47/30

    Abstract: Some implementations involve controlling a jitter buffer size during a teleconference according to a jitter buffer size estimation algorithm based, at least in part, on a cumulative distribution function (CDF). The CDF may be based, at least in part, on a network jitter parameter. The CDF may be initialized according to a parametric model. At least one parameter of the parametric model may be based, at least in part, on legacy network jitter information.

    Abstract translation: 一些实现涉及根据至少部分地基于累积分布函数（CDF）的抖动缓冲器大小估计算法在电话会议期间控制抖动缓冲器大小。 CDF可以至少部分地基于网络抖动参数。 CDF可以根据参数模型进行初始化。 参数模型的至少一个参数至少部分地基于传统网络抖动信息。

    Detecting Conference Call Performance Issue from Aberrant Behavior
    34.
    发明申请
    Detecting Conference Call Performance Issue from Aberrant Behavior 有权
    从异常行为检测电话会议问题

    公开(公告)号：US20160337510A1

    公开(公告)日：2016-11-17

    申请号：US15109511

    申请日：2015-01-06

    Applicant: Dolby Laboratories Licensing Corporation

    Inventor： Kai LI, I ,  Glenn N. DICKINS ,  Xuejing SUN

    IPC: H04M3/22 ,  H04L12/24 ,  G10L25/63 ,  H04M3/56

    CPC classification number: H04M3/2227 ,  G10L25/60 ,  G10L25/63 ,  H04L41/5009 ,  H04M3/2236 ,  H04M3/568 ,  H04M3/569 ,  H04M2203/2038

    Abstract: In a conference call having a plurality of participants interacting in a conference exchange of information in a digital transmission environment, the interaction being across a variable network transmission resource, a method of allocating the level of transmission resource, the methods including the steps of: (a) monitoring predetermined aspects of the participant's behavior during the conference call; (b) determining a divergence of participants behavior from normative values; (c) utilising any divergence as an indicator of aberrant operation of the participants; and (d) allocating the resource determinative on the divergence of participants behavior from normative values.

    Abstract translation: 在具有多个参与者在数字传输环境中的会议信息交换中进行交互的电话会议中，所述交互是跨可变网络传输资源，分配传输资源级别的方法，所述方法包括以下步骤：（ a）在电话会议期间监视参与者行为的预定方面; （b）确定参与者行为与规范价值的差异; （c）利用任何分歧作为参与者异常运作的指标; 和（d）将参与者行为与规范价值观分歧的决定因素分配。

    SPEAKER IDENTIFICATION USING SPATIAL INFORMATION
    35.
    发明申请
    SPEAKER IDENTIFICATION USING SPATIAL INFORMATION 有权
    使用空间信息的扬声器识别

    公开(公告)号：US20160180852A1

    公开(公告)日：2016-06-23

    申请号：US14971401

    申请日：2015-12-16

    Applicant: Dolby Laboratories Licensing Corporation

    Inventor： Shen HUANG ,  Xuejing SUN

    IPC: G10L17/00 ,  G10L25/24 ,  G10L25/78 ,  G10L15/30

    CPC classification number: G10L17/005 ,  G10L15/30 ,  G10L17/00 ,  G10L17/04 ,  G10L25/03 ,  G10L25/24 ,  G10L25/78 ,  H04M3/568 ,  H04M2201/41 ,  H04M2203/6045

    Abstract: Embodiments of the present invention relate to speaker identification using spatial information. A method of speaker identification for audio content being of a format based on multiple channels is disclosed. The method comprises extracting, from a first audio clip in the format, a plurality of spatial acoustic features across the multiple channels and location information, the first audio clip containing voices from a speaker, and constructing a first model for the speaker based on the spatial acoustic features and the location information, the first model indicating a characteristic of the voices from the speaker. The method further comprises identifying whether the audio content contains voices from the speaker based on the first model. Corresponding system and computer program product are also disclosed.

    Abstract translation: 本发明的实施例涉及使用空间信息的扬声器识别。 公开了一种基于多个频道的音频内容的扬声器识别方法。 该方法包括从格式的第一音频剪辑中提取多个信道上的多个空间声学特征和位置信息，所述第一音频剪辑包含来自扬声器的语音，并且基于空间来为扬声器构建第一模型 声学特征和位置信息，第一模型指示来自说话者的声音的特征。 该方法还包括基于第一模型来识别音频内容是否包含来自扬声器的语音。 还公开了相应的系统和计算机程序产品。

    ORIENTATION-AWARE SURROUND SOUND PLAYBACK
    36.
    发明申请
    ORIENTATION-AWARE SURROUND SOUND PLAYBACK 有权

    公开(公告)号：US20220264224A1

    公开(公告)日：2022-08-18

    申请号：US17736962

    申请日：2022-05-04

    Applicant: Dolby Laboratories Licensing Corporation

    Inventor： Xuejing SUN ,  Guilin MA ,  Xiguang ZHENG

    IPC: H04R5/04 ,  H04S7/00 ,  H04S1/00 ,  H04S3/00 ,  H04S3/02

    Abstract: Example embodiments disclosed herein relate to orientation-aware surround sound playback. A method for processing audio on an electronic device that includes a plurality of loudspeakers is disclosed, the loudspeakers arranged in more than one dimension of the electronic device. The method includes, responsive to receipt of a plurality of received audio streams, generating a rendering component associated with the plurality of received audio streams, determining an orientation dependent component of the rendering component, processing the rendering component by updating the orientation dependent component according to an orientation of the loudspeakers and dispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component. Corresponding system and computer program products are also disclosed.

    ADAPTIVE AUDIO FILTERING
    37.
    发明申请
    ADAPTIVE AUDIO FILTERING 审中-公开

    公开(公告)号：US20190392855A1

    公开(公告)日：2019-12-26

    申请号：US16564532

    申请日：2019-09-09

    Applicant: Dolby Laboratories Licensing Corporation

    Inventor： Dong SHI ,  Glenn N. DICKINS ,  David GUNAWAN ,  Xuejing SUN

    IPC: G10L21/0232 ,  G10L21/0208 ,  H04M9/08 ,  G10L21/028 ,  G10L25/21

    Abstract: In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal. For a frequency subband: filtered crossband references (424, 425) are formed by multiplying, by scalar factors (426, 427), filtered inband references of other subbands; a composite filtered reference (428) is formed by summing the filtered inband reference of the subband (422) and the filtered crossband references; a residual signal (429) is computed as a difference between the composite filtered reference and the subband signal of the response signal corresponding to the subband; and the scalar factors and the filter applied to the subband signal of the reference signal corresponding to the subband are adjusted based on the residual signal.

    FILTER COEFFICIENT UPDATING IN TIME DOMAIN FILTERING
    38.
    发明申请
    FILTER COEFFICIENT UPDATING IN TIME DOMAIN FILTERING 审中-公开

    公开(公告)号：US20190325891A1

    公开(公告)日：2019-10-24

    申请号：US16404611

    申请日：2019-05-06

    Applicant: Dolby Laboratories Licensing Corporation

    Inventor： Dong SHI ,  Xuejing SUN

    IPC: G10L25/18 ,  H03H21/00 ,  H03H17/02 ,  G10L25/48

    Abstract: Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a subband of the audio signal. The method also includes determining filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.

    POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT
    39.
    发明申请
    POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT 审中-公开

    公开(公告)号：US20190287548A1

    公开(公告)日：2019-09-19

    申请号：US16429552

    申请日：2019-06-03

    Applicant: Dolby Laboratories Licensing Corporation

    Inventor： Xuejing SUN ,  Glenn N. DICKINS

    IPC: G10L21/0364 ,  G10K11/16 ,  G10L21/0316 ,  H03G3/30 ,  H03G3/32 ,  G10L21/034 ,  G10L21/0224 ,  G10L25/78

    Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

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