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    • 1. 发明授权
    • DS3 to ASI converter with reduced memory
    • DS3到ASI转换器,内存减少
    • US07447231B2
    • 2008-11-04
    • US11030034
    • 2005-01-06
    • Robert S. NemiroffMichael A. CasteloesKen-Ho Chang
    • Robert S. NemiroffMichael A. CasteloesKen-Ho Chang
    • H04J3/16
    • H04N21/4307H04J3/0632H04J3/1635H04N21/23608H04N21/4305H04N21/4344
    • Converting a DS3 transport stream to MPEG-2 packets over an ASI transport stream includes extracting MPEG-2 packets along with PCR values, calculating a new PCR value to compensate for packet time shift, and replacing the extracted PCR value with the new PCR value prior to outputting MPEG-2 packets. An apparatus for processing a DS3 transport stream carrying an MPEG-2 payload includes a DS3 receiver to receive the stream and to output a data signal and clock signal. The DS3 receiver includes a DS3 interface to receive DS3 packets and discard null packets, and includes PCR correction to determine if received DS3 packets include PCR values and to adjust PCR values prior to outputting the MPEG-2 packet. Packets are output without storing each packet in a buffer, but storing the PCR value in the buffer.
    • 通过ASI传输流将DS3传输流转换为MPEG-2分组包括提取MPEG-2分组以及PCR值,计算新的PCR值以补偿分组时间偏移,并用先前的新PCR值替换提取的PCR值 输出MPEG-2数据包。 用于处理携带MPEG-2有效载荷的DS3传输流的装置包括用于接收流并输出数据信号和时钟信号的DS3接收机。 DS3接收机包括一个DS3接口,用于接收DS3数据包并丢弃零分组,并包括PCR校正,以确定接收的DS3数据包是否包括PCR值,并在输出MPEG-2数据包之前调整PCR值。 输出数据包,而不将每个数据包存储在缓冲区中,但将PCR值存储在缓冲区中。
    • 2. 发明授权
    • Audio data bit rate detector
    • 音频数据比特率检测器
    • US08510107B2
    • 2013-08-13
    • US12620762
    • 2009-11-18
    • Robert S. NemiroffJing Yang ChenSiu-Wai Wu
    • Robert S. NemiroffJing Yang ChenSiu-Wai Wu
    • G10L21/04
    • G10L19/24G10L19/167H04N21/4341H04N21/4394
    • A detector for determining an audio data bit rate of a pre-compressed audio elementary stream (AES), in which the pre-compressed AES includes a plurality of preamble length fields is provided. The detector includes an analyzer module configured to determine one or more candidate audio data bit rates of the AES from the plurality of preamble length fields, a detector module configured to determine whether one or more of the candidate audio data bit rates are correct, and a selector module configured to select one of the one or more candidate audio data bit rates determined to be correct as the audio data bit rate of the pre-compressed AES.
    • 一种用于确定预压缩音频基本流(AES)的音频数据比特率的检测器,其中预压缩AES包括多个前同步码长度字段。 检测器包括:分析器模块,被配置为从多个前导码长度字段确定AES的一个或多个候选音频数据比特率;检测器模块,被配置为确定候选音频数据比特率中的一个或多个是否正确;以及 选择器模块,被配置为选择被确定为正确的一个或多个候选音频数据比特率中的一个作为预压缩AES的音频数据比特率。
    • 4. 发明授权
    • Statistical remultiplexing with bandwidth allocation among different transcoding channels
    • 在不同代码转换通道之间进行带宽分配的统计重新复用
    • US06847656B1
    • 2005-01-25
    • US09669517
    • 2000-09-25
    • Siu-Wai WuRobert S. NemiroffVincent LiuAjay Luthra
    • Siu-Wai WuRobert S. NemiroffVincent LiuAjay Luthra
    • H04N7/26H04N7/50H04N21/236H04N21/2365H04B1/66
    • H04N21/23608H04N19/115H04N19/126H04N19/15H04N19/152H04N19/176H04N19/196H04N19/197H04N19/40H04N19/61H04N21/23655
    • An efficient statistical remultiplexer for processing a number of channels that include video data. In one aspect, transcoding of the video data is delayed while statistical information is obtained from the data. Bit rate need parameters for the data are determined based on the statistical information, and the video data is transcoded based on the respective bit rate need parameters following the delay. In another aspect, a transcoding bit rate for video frames is updated a plurality of times at successive intervals to allow a closer monitoring of the bit rate. Minimum and maximum bounds for the transcoding bit rate, e.g., for buffer overflow and underflow protection, are also updated in each interval. In another aspect, the pre-transcoding quantization scales of the macroblocks in a frame are scaled to provide corresponding new quantization scales for transcoding based on a ratio of a pre-transcoding amount of data in the frame and a target, post-transcoding amount of data for the frame. The quantization scales are adjusted for different portions of the frame as transcoding progresses to ensure that a minimum amount of transcoding bandwidth is allocated to each macroblock.
    • 一种用于处理包括视频数据的多个信道的有效统计再分集器。 在一个方面,视频数据的代码转换被延迟,同时从数据中获得统计信息。 基于统计信息确定数据的比特率需要参数,并且根据延迟后的各个比特率需求参数对视频数据进行代码转换。 在另一方面,视频帧的代码转换比特率以连续的间隔多次更新,以允许更仔细地监视比特率。 代码转换比特率的最小和最大限制,例如用于缓冲区溢出和下溢保护,也在每个间隔中被更新。 在另一方面,对帧中的宏块进行预先转码的量化尺度进行缩放,以提供相应的新的用于代码转换的量化尺度,其基于帧中的数据的预转码量与目标之间的比率,后转码量 帧的数据。 当代码转换进行时,针对帧的不同部分调整量化尺度,以确保将最小量的代码转换带宽分配给每个宏块。
    • 8. 发明授权
    • Audio splice process for digital Ad insertion
    • 用于数字广告插入的音频拼接过程
    • US07725202B2
    • 2010-05-25
    • US10744271
    • 2003-12-23
    • Jingyang ChenRobert S. Nemiroff
    • Jingyang ChenRobert S. Nemiroff
    • G06F17/00H04N9/74
    • H04N21/2335H04N21/23424H04N21/44016H04N21/8106H04N21/812
    • A system and method for audio splicing (insertion) of an Ad audio stream in the compressed domain, where variable early delivery of the Ad audio stream and variable bit rate are allowed, without creating audio distortion, glitches, or other digital artefacts or errors, in the resultant audio stream is disclosed. The present system and method provides for a splice delay buffer which delays the first five Ad audio frames until transmission of the last frame of the primary audio stream, but before the splice time. Subsequent Ad audio frames are delayed by a fixed amount, where the fixed amount is greater than the frame delay of the primary audio stream, to allow for ease of splice back to the primary audio stream.
    • 用于音频拼接(插入)压缩域中的Ad音频流的系统和方法,其中允许可变的早期传送Ad音频流和可变比特率,而不产生音频失真,毛刺或其他数字伪像或错误, 在所得到的音频流中被公开。 本系统和方法提供了一种拼接延迟缓冲器,其将前五个Ad音频帧延迟,直到主音频流的最后一帧传输,但在拼接时间之前。 随后的广告音频帧被延迟固定量,其中固定量大于主音频流的帧延迟,以允许容易地将主音频流连接起来。