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公开(公告)号：KR101808164B1
公开(公告)日：2017-12-13
申请号：KR1020150179547
申请日：2015-12-15
Applicant: 한국전자통신연구원
IPC: H04L1/00
Abstract: 본발명의실시예들은, 패킷네트워크상에서손실이발생된오디오패킷을복구하기위한방법및 장치에관한것으로, 본발명의일 실시예에따른오디오패킷전송장치가오디오패킷손실을복구하는방법은, 선형독립인랜덤이진행렬과 HDPC(High Density Parity Check) 행렬을이용하여프리코딩행렬을생성하는단계; 원본심볼과상기프리코딩행렬의역행렬을이용하여중간심볼들을생성하는단계; 및상기중간심볼들과디그리(degree) 분포행렬을이용하여인코딩된원본심볼및 인코딩된복구심볼을생성하는단계를포함한다. 본발명의실시예들에따르면, 작은원본심볼블록처리를위한저지연 FEC 코드를적용함으로써영상회의를위한종단간오디오지연시간을줄일수 있다.
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公开(公告)号：KR1020150079367A
公开(公告)日：2015-07-08
申请号：KR1020140051262
申请日：2014-04-29
Applicant: 한국전자통신연구원
CPC classification number: H04N7/147 , H04L65/403 , H04N7/152 , H04N19/10 , H04N19/44 , H04N21/4788
Abstract: 본발명은영상데이터통신방법에관한것이다. 본발명의영상데이터통신방법은, 영상데이터를인코딩하여인코딩데이터를생성하는단계, 인코딩데이터를이용하여복구데이터를생성하는단계, 인코딩데이터및 복구데이터를패킷화하여패킷들을생성하는단계, 그리고생성된패킷들을전송하는단계로구성된다. 복구데이터를생성하는단계는, 인코딩데이터를복수의조각데이터들로분할하는단계, 그리고복수의조각데이터들중 적어도두 개의조각데이터들에대해배타적논리합연산을수행하여, 복구데이터를생성하는단계로구성된다.
Abstract translation: 图像数据通信方法技术领域本发明涉及一种图像数据通信方法。 根据本发明，图像数据传输方法包括以下步骤：通过编码图像数据生成编码数据; 通过使用编码数据生成恢复数据; 打包编码数据和恢复数据以生成数据包; 并发送生成的分组。 恢复数据生成步骤包括以下步骤：将编码数据分割成多个数据片段; 并且用两个或更多个数据片段执行异或运算以产生恢复数据。
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公开(公告)号：KR1020140011514A
公开(公告)日：2014-01-29
申请号：KR1020120067560
申请日：2012-06-22
Applicant: 한국전자통신연구원
CPC classification number: H04L5/0044 , H04L1/0047 , H04L43/0829 , H04L69/22
Abstract: The present invention relates to an apparatus and a method for normal communicating data by recovering a loss of data packets to enhance multimedia service quality in a wireless communication based all-internet protocol (all-IP) network. The method for communication data comprises steps of receiving receipt information of the data packet on a receiver from the receiver; generating a plurality of frames including the data to be transmitted to the receiver by considering the receipt information, including the frames in the payload and generating the data packets including a header and the payload; and transmitting data packets to the receiver wherein the payload includes the frames and a mode field with information on the frames, and the mode field includes the received information and information for indicating the number of frames.
Abstract translation: 本发明涉及通过在无线通信的全网协议（全IP）网络中恢复丢失数据分组以增强多媒体业务质量来正常传送数据的装置和方法。 通信数据的方法包括以下步骤：从接收机在接收机上接收数据分组的接收信息; 通过考虑包括有效载荷中的帧并且生成包括报头和有效载荷的数据分组的接收信息来生成包括要发送到接收机的数据的多个帧; 以及向所述接收机发送数据分组，其中所述有效载荷包括所述帧，以及具有关于所述帧的信息的模式字段，并且所述模式字段包括所接收的信息和用于指示所述帧数的信息。
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公开(公告)号：KR1020110070459A
公开(公告)日：2011-06-24
申请号：KR1020090127288
申请日：2009-12-18
Applicant: 한국전자통신연구원
Abstract: PURPOSE: A codec bit rate control method in handover between heterogeneous networks is provided to control a mode corresponding to the bit rate of multimedia codec or a bit rate on the basis of the transmission bandwidth of a target network in the inter-heterogeneous network handover of a portable terminal supporting multi-modes. CONSTITUTION: A portable terminal detects handover(310). The portable terminal compares the transmission bandwidth of a target network with the transmission bandwidth of a present network(320). In case the transmission bandwidth of the target network is greater than the transmission bandwidth of the present network, the portable terminal performs the handover(330). The portable terminal changes the bit rate of codec(340). In case the transmission bandwidth of the target network is smaller than the transmission bandwidth of the present network, the portable terminal firstly changes the bit rate of codec(350).
Abstract translation: 目的：提供异构网络之间的切换中的编解码器比特率控制方法，以在异构网络切换中的目标网络的传输带宽上控制与多媒体编解码器的比特率相对应的模式或比特率 便携式终端支持多模式。 构成：便携式终端检测到切换（310）。 便携式终端将目标网络的传输带宽与当前网络的传输带宽进行比较（320）。 在目标网络的传输带宽大于当前网络的传输带宽的情况下，便携式终端执行切换（330）。 便携式终端改变编解码器的比特率（340）。 在目标网络的传输带宽小于当前网络的传输带宽的情况下，便携式终端首先改变编解码器的比特率（350）。
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公开(公告)号：KR100875936B1
公开(公告)日：2008-12-26
申请号：KR1020070067187
申请日：2007-07-04
Applicant: 한국전자통신연구원
Abstract: A method and an apparatus for matching measurement interval of speech quality of variable band multi-codec are provided to match a measurement interval between a reference sound and a codec sound and reduce a measurement result distortion phenomenon. A measurement interval setting unit(110) sets an interval of speech quality desired to be measured by time intervals, and the first conversion unit(120) converts the measurement interval by sample units. An information synthesizing unit(130) synthesizes information about the measurement interval which has been converted by sample units and a digital original sound, and outputs the same.
Abstract translation: 提供了一种用于匹配可变频带多编解码器的语音质量的测量间隔的方法和设备，以匹配参考声音和编解码器声音之间的测量间隔并减少测量结果失真现象。 测量间隔设置单元（110）按时间间隔设置期望测量的语音质量的间隔，并且第一转换单元（120）将测量间隔转换为采样单位。 信息合成单元（130）合成关于已被样本单元转换的测量间隔的信息和数字原始声音，并将其输出。
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Patent Agency Ranking
公开(公告)号：KR1020080052258A
公开(公告)日：2008-06-11
申请号：KR1020070067030
申请日：2007-07-04
Applicant: 한국전자통신연구원
Abstract: A method for controlling a variable bit-rate voice codec, and an apparatus thereof are provided to ensure QoS(Quality of Service) of terminal service and reduce loads of a codec by using a single codec which supports a variable bit-rate. An apparatus(20) for controlling a variable bit-rate voice codec comprises a terminal information extraction unit(21), a call quality monitoring unit(22) and an admission policy unit. The terminal information extraction unit extracts telephone call capability of terminals which require new call set up. The call quality monitoring unit extracts network parameters from existing calls through exchanged packets, and measures the voice quality of the existing calls on the basis of the network parameters. The admission policy unit decides the admission of the new call on the basis of the telephone call capability and the voice quality.
Abstract translation: 提供一种用于控制可变比特率语音编解码器的方法及其装置，以确保终端服务的QoS（服务质量），并通过使用支持可变比特率的单个编解码器来减少编解码器的负担。 用于控制可变比特率语音编解码器的装置（20）包括终端信息提取单元（21），呼叫质量监视单元（22）和准入策略单元。 终端信息提取单元提取需要新呼叫建立的终端的电话呼叫能力。 呼叫质量监控单元通过交换的分组从现有呼叫中提取网络参数，并根据网络参数测量现有呼叫的语音质量。 接入策略单元根据电话呼叫能力和语音质量来决定新呼叫的接纳。
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公开(公告)号：KR100833499B1
公开(公告)日：2008-05-29
申请号：KR1020060104789
申请日：2006-10-27
Applicant: 한국전자통신연구원
Abstract: 가변 대역 멀티 코덱의 음성 품질 측정 장치 및 방법이 개시된다. 본 발명은 자연음을 녹음한 제 1 녹음파일, 및 자연음을 코덱을 이용하여 디지털 데이터로 변환시킨 제 2 녹음파일을 수신하고, 자연음을 코덱으로 인코딩한 정보를 RTP 패킷 형태로 수신한 후 RTP 패킷을 해체하고 코덱으로 디코딩하여 제 3 녹음파일을 생성하는 녹음파일 수신/생성부; 제 1 녹음파일 내지 제 3 녹음파일에서 하나의 파일을 중복 선택하거나 두 개의 파일을 선택하여, 서로 상이한 정도를 MOS 값으로 산출하는 MOS 값 산출부; 및 MOS 값 생성부에서 생성된 MOS 값들을 비교하여 음성 품질 왜곡 원인을 파악하는 MOS 값 비교부;를 포함한다. 본 발명에 따르면, 가변 대역 멀티 코덱의 단대단 음성 품질 측정을 더욱 정확하게 할 수 있으며, 자연음 왜곡 등 음성 품질이 저하되는 현상의 원인을 파악할 수 있다.
품질 측정, VoIP, 코덱, 단대단, End-to-End, 가변 대역, QoS-
公开(公告)号：KR100744567B1
公开(公告)日：2007-08-01
申请号：KR1020060096457
申请日：2006-09-29
Applicant: 한국전자통신연구원
Abstract: An apparatus for minimizing the number of transcoding times in a multi-network multi-CODEC(Coder Decoder) environment and a method thereof are provided to select a CODEC having the smallest number of transcoding times by delivering transcoding information, thereby minimizing the number of transcoding times, consequently a factor of deteriorating the quality of original media can be minimized. A receiving device receives a CODEC of a sending device, and the number of times the corresponding CODEC is transcoded from a CODEC of the first sending terminal(410). The receiving device records the received CODEC and the number of the transcoding times of the corresponding CODEC to prepare receiving CODEC lists(420). The receiving device perceives whether the self is the final receiving device(430). If so, the receiving device determines a CODEC having high priority as a response CODEC among the receiving CODEC lists(440).
Abstract translation: 提供了一种用于最小化多网络多编解码器（编码解码器）环境中的转码时间的数量的设备及其方法，以通过递送转码信息来选择具有最少的转码时间的编解码器，由此最小化转码的数量 次，因此可以最小化原始媒体质量恶化的因素。 接收设备接收发送设备的CODEC，并且从第一发送终端（410）的CODEC转码对应的CODEC的次数。 接收设备记录所接收的编解码器和相应编解码器的编码转换时间的数量以准备接收编解码器列表（420）。 接收设备感知自己是否是最终的接收设备（430）。 如果是，则接收设备在接收CODEC列表中确定具有高优先级的CODEC作为响应CODEC（440）。
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