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    • 2. 发明申请
    • Measurement of round-trip delay over a network
    • 测量网络上的往返延迟
    • US20080056154A1
    • 2008-03-06
    • US11516933
    • 2006-09-06
    • Scott FirestoneRandall B. BairdWing Cheong ChauShantanu Sarkar
    • Scott FirestoneRandall B. BairdWing Cheong ChauShantanu Sarkar
    • H04J1/16H04L12/66
    • H04L12/66
    • In one embodiment, a first audio waveform is produced at a first side of a network connection and then encoded and sent by a first endpoint device to a second endpoint device at a second side of the network connection. A second audio waveform is then detected after being played out by the first endpoint device, the second audio waveform having been produced at the second side of the network connection in response to the second endpoint device playing out the first audio waveform. A round-trip delay is then calculating based on a time period measured from output of the first audio waveform to detection of the second audio waveform. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure.
    • 在一个实施例中,在网络连接的第一侧产生第一音频波形,然后由第一端点设备编码并发送到网络连接的第二侧的第二端点设备。 然后在由第一端点设备播放之后检测第二音频波形,响应于第二端点设备播放第一音频波形,已经在网络连接的第二侧产生了第二音频波形。 然后基于从第一音频波形的输出测量到第二音频波形的检测的时间周期来计算往返延迟。 要强调的是,该摘要被提供以符合要求抽象的规则,允许搜索者或其他读者快速确定技术公开内容的主题。
    • 6. 发明授权
    • Non-causal speaker selection for conference multicast
    • 会议组播的非因果音箱选择
    • US06963353B1
    • 2005-11-08
    • US10439147
    • 2003-05-14
    • Scott Firestone
    • Scott Firestone
    • H04N7/14H04N7/15
    • H04N7/147H04N7/152
    • A method for non-causal speaker selection is provided. In accordance with a particular embodiment of the present invention the method includes receiving a plurality of video streams at a multipoint control unit, each of the plurality of video streams being associated with a respective endpoint of a multipoint conference. A plurality of audio streams may also be received at the multipoint control unit, and each audio stream may be associated with a respective one of the video streams. The audio streams are buffered in respective audio buffers, and the video streams are buffered in respective video buffers. First video data is copied from the video buffers to obtain a low latency video stream for distribution to active conference participants. In a particular embodiment, second video data may be copied from the video buffers to obtain a high latency video stream for distribution to passive conference participants, the high latency video streams being delayed in time with respect to the low latency video stream.
    • 提供了非因果说话者选择的方法。 根据本发明的特定实施例,该方法包括在多点控制单元处接收多个视频流,多个视频流中的每一个与多点会议的相应端点相关联。 也可以在多点控制单元处接收多个音频流,并且每个音频流可以与视频流中的相应一个相关联。 音频流被缓冲在相应的音频缓冲器中,并且视频流被缓冲在相应的视频缓冲器中。 从视频缓冲器复制第一视频数据,以获得低延迟视频流,以便分发给活动的会议参与者。 在特定实施例中,可以从视频缓冲器复制第二视频数据以获得用于分配给被动会议参与者的高等待时间视频流,相对于低等待时间视频流,高等待时间视频流在时间上被延迟。
    • 7. 发明授权
    • Real-time data rate matching across a medium
    • 跨媒介的实时数据速率匹配
    • US06247072B1
    • 2001-06-12
    • US09013866
    • 1998-01-27
    • Scott Firestone
    • Scott Firestone
    • G06F1300
    • H04N21/431G06F5/06G06F5/12G06F2205/061G06F2205/126H04N21/23406H04N21/2381H04N21/4307H04N21/44004H04N21/440281H04N21/6437
    • Apparatus and methods for matching data rates is useful for a receiver receiving real-time data over a medium. Implementations feature a process establishing a buffer in a receiver; receiving source data from a source having a nominal source data rate, the received source data arriving at an incoming data rate that differs from time-to-time from the nominal source data rate; filling the buffer with source data as it is received at the incoming data rate and emptying the buffer to provide data for consumption in real time at a consumption data rate; setting a rate-matching factor M, the factor M affecting the rate at which the buffer is emptied; and tracking the level of data in the buffer and resetting the value of M to increase the rate at which the buffer is emptied when the buffer fills above a target range, and resetting the value of M to decrease the rate at which the buffer is emptied when the buffer empties below a target range.
    • 用于匹配数据速率的装置和方法对于在介质上接收实时数据的接收器是有用的。 实现特征在于在接收器中建立缓冲器的过程; 从具有标称源数据速率的源接收源数据,所接收的源数据以与标称源数据速率的时间不同的输入数据速率到达; 在传入数据速率接收到源数据时填充缓冲器,并将缓冲器清空以消耗数据速率实时提供消耗数据; 设置速率匹配因子M,影响缓冲器清空速率的因子M; 并且跟踪缓冲器中的数据级别并重置M的值以在缓冲器填满目标范围时提高缓冲器被清空的速率,并且重置M的值以降低缓冲器被清空的速率 当缓冲液清空到目标范围以下时。
    • 8. 发明授权
    • Measurement of round-trip delay over a network
    • 测量网络上的往返延迟
    • US07916653B2
    • 2011-03-29
    • US11516933
    • 2006-09-06
    • Scott FirestoneRandall B. BairdWing Cheong ChauShantanu Sarkar
    • Scott FirestoneRandall B. BairdWing Cheong ChauShantanu Sarkar
    • G01R31/08H04L12/66
    • H04L12/66
    • In one embodiment, a first audio waveform is produced at a first side of a network connection and then encoded and sent by a first endpoint device to a second endpoint device at a second side of the network connection. A second audio waveform is then detected after being played out by the first endpoint device, the second audio waveform having been produced at the second side of the network connection in response to the second endpoint device playing out the first audio waveform. A round-trip delay is then calculating based on a time period measured from output of the first audio waveform to detection of the second audio waveform. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure.
    • 在一个实施例中,在网络连接的第一侧产生第一音频波形,然后由第一端点设备编码并发送到网络连接的第二侧的第二端点设备。 然后在由第一端点设备播放之后检测第二音频波形,响应于第二端点设备播放第一音频波形,已经在网络连接的第二侧产生了第二音频波形。 然后基于从第一音频波形的输出测量到第二音频波形的检测的时间周期来计算往返延迟。 要强调的是,该摘要被提供以符合要求抽象的规则,允许搜索者或其他读者快速确定技术公开内容的主题。