ADAPTIVE AUDIO STREAM WITH LATENCY COMPENSATION
    1.
    发明申请
    ADAPTIVE AUDIO STREAM WITH LATENCY COMPENSATION 有权
    自适应音频流

    公开(公告)号:US20160105473A1

    公开(公告)日:2016-04-14

    申请号:US14581321

    申请日:2014-12-23

    Abstract: A latency compensating adaptive jitter buffer method is implemented in an audio client running on a user device or in an audio server to adaptively adjust the size of a jitter buffer to optimize latency while minimizing packet loss during audio signal transmission. In some embodiments, the jitter buffer is kept to a nominal size for low latency. In response to a queuing delay event being detected, audio production is temporarily stopped and the size of the jitter buffer is temporarily increased to receive all incoming audio packets up to a certain refill level. The method then resumes audio production using accelerated playback to reduce the jitter buffer size back to the nominal size.

    Abstract translation: 延迟补偿自适应抖动缓冲方法在运行在用户设备或音频服务器中的音频客户端中实现,以自适应地调整抖动缓冲器的大小以优化等待时间,同时最小化音频信号传输期间的分组丢失。 在一些实施例中,抖动缓冲器保持在标称大小以用于低等待时间。 响应于检测到排队延迟事件,音频产生被暂时停止,并且暂时增加抖动缓冲器的大小以接收所有输入音频分组,直到一定的再充值电平。 然后,该方法使用加速重放来恢复音频制作,以将抖动缓冲器大小减小到标称大小。

    DYNAMIC LOCALE BASED AGGREGATION OF FULL DUPLEX MEDIA STREAMS
    2.
    发明申请
    DYNAMIC LOCALE BASED AGGREGATION OF FULL DUPLEX MEDIA STREAMS 有权
    基于动态LOCALE的全双工媒体流的聚合

    公开(公告)号:US20160014373A1

    公开(公告)日:2016-01-14

    申请号:US14580866

    申请日:2014-12-23

    Abstract: A cloud-based video/audio conferencing system and method performs locale based aggregation of a full duplex media stream to organize multiple connections to a conference call that originate from the same physical location or a shared locale. The cloud-based video/audio conferencing system performs synchronization of the microphone and speaker audio signals of the same-locale connections. In this manner, a conference call may be held with multiple user devices making connections from the same physical location. User experience is enhanced by allowing each user in the same location to use his/her own individual devices to connect to the same conference call

    Abstract translation: 基于云的视频/音频会议系统和方法执行全双工媒体流的基于区域设置的聚合,以组织来自相同物理位置或共享语言环境的电话会议的多个连接。 基于云的视频/音频会议系统执行同一地点连接的麦克风和扬声器音频信号的同步。 以这种方式,可以与多个用户设备进行电话会议,从相同的物理位置进行连接。 通过允许同一位置的每个用户使用自己的个人设备连接到同一个电话会议来增强用户体验

    Adaptive audio stream with latency compensation
    3.
    发明授权
    Adaptive audio stream with latency compensation 有权
    具有延迟补偿的自适应音频流

    公开(公告)号:US09521057B2

    公开(公告)日:2016-12-13

    申请号:US14581321

    申请日:2014-12-23

    Abstract: A latency compensating adaptive jitter buffer method is implemented in an audio client running on a user device or in an audio server to adaptively adjust the size of a jitter buffer to optimize latency while minimizing packet loss during audio signal transmission. In some embodiments, the jitter buffer is kept to a nominal size for low latency. In response to a queuing delay event being detected, audio production is temporarily stopped and the size of the jitter buffer is temporarily increased to receive all incoming audio packets up to a certain refill level. The method then resumes audio production using accelerated playback to reduce the jitter buffer size back to the nominal size.

    Abstract translation: 延迟补偿自适应抖动缓冲方法在运行在用户设备或音频服务器中的音频客户端中实现,以自适应地调整抖动缓冲器的大小以优化等待时间,同时最小化音频信号传输期间的分组丢失。 在一些实施例中,抖动缓冲器保持在标称大小以用于低等待时间。 响应于检测到排队延迟事件,音频产生被暂时停止,并且暂时增加抖动缓冲器的大小以接收所有输入音频分组,直到一定的再充值电平。 然后,该方法使用加速重放来恢复音频制作,以将抖动缓冲器大小减小到标称大小。

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