Controlling a jitter buffer
    31.
    发明授权

    公开(公告)号:US10560393B2

    公开(公告)日:2020-02-11

    申请号:US14654346

    申请日:2013-12-19

    Abstract: Apparatus and methods for controlling a jitter buffer are described. In one embodiment, the apparatus for controlling a jitter buffer includes an inter-talkspurt delay jitter estimator for estimating an offset value of the delay of a first frame in the current talkspurt with respect to the delay of a latest anchor frame in a previous talkspurt, and a jitter buffer controller for adjusting a length of the jitter buffer based on a long term length of the jitter buffer for each frame and the offset value.

    Audio spatial rendering apparatus and method

    公开(公告)号:US09854378B2

    公开(公告)日:2017-12-26

    申请号:US14768676

    申请日:2014-01-30

    CPC classification number: H04S7/30 H04S2400/11 H04S2420/01

    Abstract: An audio spatial rendering apparatus and method are disclosed. In one embodiment, The audio spatial rendering apparatus includes a rendering unit for spatially rendering an audio stream so that the reproduced far-end sound is perceived by a listener as originating from at least one virtual spatial position, a real position obtaining unit for obtaining a real spatial position of a real sound source, a comparator for comparing the real spatial position with the at least one virtual spatial position; and an adjusting unit for, where the real spatial position is within a predetermined range around at least one virtual spatial position, or vice versa, adjusting the parameters of the rendering unit so that the at least one virtual spatial position is changed.

    Near Optimal Forward Error Correction System and Method

    公开(公告)号:US20170104552A1

    公开(公告)日:2017-04-13

    申请号:US15287868

    申请日:2016-10-07

    Abstract: A method of determining a near optimal forward error correction scheme for the transmission of audio data over a lossy packet switched network having preallocated estimated bandwidth, delay and packet losses, between at least a first and second communications devices, the method including the steps of: determining a first coding rate for the audio data; determining a peak redundancy coding rate for redundant versions of the audio data; determining an average redundancy coding rate over a period of time for redundant versions of the audio data; determining an objective function which maximizes a bitrate-perceptual audio quality mapping of the transmitted audio data including a playout function formulation; and optimising the objective function to produce a forward error correction scheme providing a high bitrate perceptual audio quality.

    Position-dependent hybrid domain packet loss concealment
    37.
    发明授权
    Position-dependent hybrid domain packet loss concealment 有权
    位置相关混合域丢包隐藏

    公开(公告)号:US09514755B2

    公开(公告)日:2016-12-06

    申请号:US14431256

    申请日:2013-09-27

    CPC classification number: G10L19/005 G10L19/0017

    Abstract: The present document relates to audio signal processing in general, and to the concealment of artifacts that result from loss of audio packets during audio transmission over a packet-switched network, in particular. A method (200) for concealing one or more consecutive lost packets is described. A lost packet is a packet which is deemed to be lost transform-based audio decoder. Each of the one or more lost packets comprises a set of transform coefficients. A set of transform coefficients is used by the transform-based audio decoder to generate a corresponding frame of a time domain audio signal. The method (200) comprises determining (205) for a current lost packet of the one or more lost packets a number of preceding lost packets from the one or more lost packets; wherein the determined number is referred to as a loss position. Furthermore, the method comprises determining a packet loss concealment, referred to as PLC, scheme based on the loss position of the current packet; and determining (204, 207, 208) an estimate of a current frame of the audio signal using the determined PLC scheme (204, 207, 208); wherein the current frame corresponds to the current lost packet.

    Abstract translation: 本文件一般涉及音频信号处理,特别涉及在通过分组交换网络的音频传输期间由于音频分组丢失而导致的伪影的隐藏。 描述用于隐藏一个或多个连续丢失分组的方法(200)。 丢失的分组是被认为是丢失的基于变换的音频解码器的分组。 一个或多个丢失分组中的每一个包括一组变换系数。 基于变换的音频解码器使用一组变换系数来生成时域音频信号的相应帧。 所述方法(200)包括:从所述一个或多个丢失分组确定(205)所述一个或多个丢失分组的当前丢失分组的若干先前丢失分组; 其中所确定的数量被称为损失位置。 此外,该方法包括基于当前分组的丢失位置确定称为PLC的分组丢失隐藏; 以及使用所确定的所述PLC方案(204,207,208)确定所述音频信号的当前帧的估计(204,207,208); 其中当前帧对应于当前丢失分组。

    Audio Signal Processing
    38.
    发明申请
    Audio Signal Processing 审中-公开
    音频信号处理

    公开(公告)号:US20150254054A1

    公开(公告)日:2015-09-10

    申请号:US14630869

    申请日:2015-02-25

    CPC classification number: G06F3/165 G05B15/02

    Abstract: A method for audio signal processing is provided. The method includes acquiring a first set of metadata associated with consumption of an audio signal by a target user, acquiring a second set of metadata associated with a set of reference users and generating, at least partially based on the first and second sets of metadata, a recommended configuration of at least one parameter for the target user, the at least one parameter being for use in the consumption of the audio signal. Corresponding apparatus and computer program product are also disclosed.

    Abstract translation: 提供了一种用于音频信号处理的方法。 该方法包括获取由目标用户消费音频信号相关联的第一组元数据,获取与一组参考用户相关联的第二组元数据,并至少部分地基于第一和第二组元数据, 用于目标用户的至少一个参数的推荐配置,该至少一个参数用于消费音频信号。 还公开了相应的设备和计算机程序产品。

    POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT
    39.
    发明申请
    POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT 有权
    后处理增益信号增强

    公开(公告)号:US20150030180A1

    公开(公告)日:2015-01-29

    申请号:US14384372

    申请日:2013-03-21

    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.

    Abstract translation: 一种用于后处理由输入处理确定的原始增益以产生后处理增益的方法,装置和逻辑,包括使用增量增益平滑和决策导向增益平滑中的一者或两者。 Δ增益平滑包括以平滑因子对原始增益应用平滑滤波器,该平滑因子取决于增益增量:当前帧的原始增益与前一帧的后处理增益之间的差的绝对值。 决策导向增益平滑包括将原始增益转换为信噪比,将具有平滑因子的平滑滤波器应用于信噪比以计算平滑的信噪比,并将 平滑的信噪比以确定第二平滑增益,平滑因子可能取决于增益增量。

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