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公开(公告)号：KR1020070059834A
公开(公告)日：2007-06-12
申请号：KR1020060041696
申请日：2006-05-09
Applicant: 한국전자통신연구원 , 한국정보통신대학교 산학협력단 , 주식회사 케이티
CPC classification number: H04L43/0829 , H04M7/006
Abstract: A network-based PLC(Packet Loss Concealment) method and an apparatus for the same are provided to maintain the quality of VoIP services regularly by increasing a PLC effect in a lost voice packet interval as a receiving unit executes PLC with loss-concealed voice packets having high correlation. A PLC apparatus(320) comprises a packet receiving part(321), a PLC processing part(322), a history buffer(323), and a packet transmitting part(324). The packet receiving part(321) receives VoIP packets, transmitted from a transmitting unit. The PLC processing part(322) judges whether packet loss has occurred in the VoIP packets received from the packet receiving part(321), and performs packet loss compensation for lost packets. The history buffer(323) stores the VoIP packets received from the PLC processing part(322). The packet transmitting part(324) forwards the VoIP packets, received from the PLC processing part(322), to the final receiving unit through a network.
Abstract translation: 提供了一种基于网络的PLC（Packet Loss Concealment，分组丢包隐藏）方法及其装置，用于通过在丢失语音分组间隔中增加PLC效应来定期维护VoIP服务的质量，因为接收单元执行具有丢失隐藏语音分组的PLC 具有很高的相关性。 PLC装置（320）包括分组接收部分（321），PLC处理部分（322），历史缓冲器（323）和分组发送部分（324）。 分组接收部分（321）从发送单元接收VoIP分组。 PLC处理部（322）判断从分组接收部（321）接收到的VoIP分组中是否发生分组丢失，并对丢包进行分组丢失补偿。 历史缓冲器（323）存储从PLC处理部（322）接收的VoIP分组。 分组发送部（324）通过网络将从PLC处理部（322）接收的VoIP分组转发到最终的接收部。
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公开(公告)号：KR1020100071803A
公开(公告)日：2010-06-29
申请号：KR1020080130644
申请日：2008-12-19
IPC: H04N19/89
CPC classification number: H04N19/895
Abstract: PURPOSE: A method for restoring a transport error in an image and an apparatus thereof are provided to efficiently restore error generated when an image is transmitted under a communication environment, thereby restoring an original image. CONSTITUTION: A discriminating unit(20) obtains the standard deviation of pixel values about image blocks of each damaged block. The discriminating unit determines the complexity of a corresponding image block based on the obtained standard deviation. An error restoring unit(30) selects one of the entire pixel information and regional pixel information based on the complexity discriminated for each image block. The error restoring unit restores an error area included in a corresponding image block based on the selected information.
Abstract translation: 目的：提供一种用于恢复图像中的传输错误的方法及其装置，以有效地恢复在通信环境下传输图像时产生的错误，从而恢复原始图像。 构成：识别单元（20）获得关于每个损坏块的图像块的像素值的标准偏差。 鉴别单元基于获得的标准偏差确定相应图像块的复杂度。 错误恢复单元（30）基于为每个图像块识别的复杂度来选择整个像素信息和区域像素信息之一。 错误恢复单元基于所选择的信息恢复包括在相应图像块中的错误区域。
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公开(公告)号：KR100922898B1
公开(公告)日：2009-10-20
申请号：KR1020070132442
申请日：2007-12-17
CPC classification number: H04L41/509 , H04L41/5009
Abstract: 본 발명은 IP 네트워크 기반의 미디어 서비스에서 제공되는 영상의 품질을 측정하기 위한 장치 및 방법에 관한 것으로서, 더욱 상세하게는 전송영상과 원본영상을 수신하여 프레임을 추출하고, 추출된 프레임을 하위 블록으로 분할하여 휘도에 따른 랭크맵을 작성한 후, 프레임 간의 거리를 계산하여 원본영상과 매칭(matching)되는 전송영상의 특정구간을 검출하는 것을 특징으로 한다. 이렇게 검출된 전송영상의 특정구간과 원본영상을 이용하여 영상의 QoE 파라미터를 측정하면, 실시간으로 영상의 품질을 모니터링할 수 있기 때문에 보다 효율적으로 영상 품질을 측정할 수 있다.
전IP화(All-IP), IP 미디어(IP-media), QoE, 영상품질, 랭크맵-
公开(公告)号：KR100847138B1
公开(公告)日：2008-07-18
申请号：KR1020060050906
申请日：2006-06-07
Abstract: 본 발명은 VoIP(Voice over Internet Protocol) 망에서 서비스 품질이 보장된 통화 음질 제공을 위한 통화 음질 측정 방법 및 시스템에 관한 것으로, 본 발명의 VoIP 망에서 통화 음질 측정 시스템은, IP를 이용한 패킷 전송을 통해 송수신 단 간에 음성 통화 서비스를 제공하는 음성통화망, 상기 송수신 단 간에 호가 설정된 상태에서 수신되는 통화음질측정 요구신호에 대해 참조음성파일을 생성하여 전송하고 상기 생성한 참조음성파일을 패킷화하여 시험음성데이터로 변환하여 전송하는 음질측정서버, 및 입력되는 명령에 따라 상기 통화음질측정 요구신호를 상기 음질측정서버로 전송하고 상기 음질측정서버로부터 전송된 참조음성파일을 기준으로 상기 시험음성데이터의 왜곡에 따라 발생하는 통화음질 저해요소와 기 저장된 통화음질 저해요소의 값을 비교하여 산출되는 편차 값에 기초해 상기 음성통화망의 장애에 의한 통화음질 저해요소를 제거하여 서비스 품질이 보장된 통화 음질 서비스를 제공하는 소프트폰 단말을 포함하며, 이에 따라, 보다 서비스 품질(QoS)이 좋은 음성통화품질을 제공할 수 있고 통화 품질 유지 보수를 위한 업무 효율을 증대시킬 수 있다.
VoIP, PESQ, 통화 음질, 품질, 측정, QoS, 패킷, 음성-
公开(公告)号：KR100776797B1
公开(公告)日：2007-11-19
申请号：KR1020060041696
申请日：2006-05-09
Applicant: 한국전자통신연구원 , 한국정보통신대학교 산학협력단 , 주식회사 케이티
Abstract: 본 발명은 네트워크 기반의 패킷 손실 은닉 방법 및 이를 위한 장치에 관한 것으로서, 기존의 수신단에서만 패킷 손실 은닉 방법에 의하여 손실된 음성 패킷을 은닉하는 경우에 비해, 네트워크상에서 손실된 패킷과 상관도가 높은 음성 패킷을 가지고 수신단에서 패킷 손실 은닉을 미리 수행함으로써, 손실된 음성 패킷 구간에 대하여 패킷 손실 은닉 효과를 더욱 향상시킬 수 있어 VoIP 서비스의 품질을 일정하게 유지할 수 있도록 한 것이다.
패킷 손실 은닉, VoIP, 히스토리 버퍼, PLC-
Patent Agency Ranking
公开(公告)号：KR1020070060861A
公开(公告)日：2007-06-13
申请号：KR1020050120973
申请日：2005-12-09
Abstract: A method for measuring bi-directional interactive speech quality in a VoIP(Voice over Internet Protocol) network is provided to provide a high-quality VoIP service to a user by predicting accurate speech quality between ends. When measurement of speech quality is requested, a test VoIP packet for measuring speech quality is generated and transmitted to the other party(S41). A transmission side and a reception side receive the test VoIP packet and analyze it(S42). A damage value(Ie) caused by a packet loss is calculated through a PESQ(Perceptual Evaluation of Speech Quality) algorithm(S43). A delay value(Id) in accordance with an end-to-end packet delay is calculated based on an actual VoIP network(S44). A jitter value(Ij) in accordance with irregularity of the end-to-end packet delay is calculated based on the actual VoIP network(S45). A speech quality measurement value between a transmitter and a receiver is calculated by applying the damage value(Ie), the delay value(Id) and the jitter value(Ij) to a new E-Model(S46).
Abstract translation: 提供了一种用于测量VoIP（因特网协议语音）网络中的双向交互语音质量的方法，通过预测终端之间的准确语音质量向用户提供高质量的VoIP服务。 当请求语音质量的测量时，生成用于测量语音质量的测试VoIP分组并发送给另一方（S41）。 发送侧和接收侧接收测试VoIP分组并对其进行分析（S42）。 通过PESQ（语音质量感知评估）算法计算由分组丢失引起的损伤值（Ie）（S43）。 基于实际的VoIP网络计算出根据端到端分组延迟的延迟值（Id）（S44）。 根据实际的VoIP网络计算根据端到端分组延迟的不规则性的抖动值（Ij）（S45）。 通过将损伤值（Ie），延迟值（Id）和抖动值（Ij）应用于新的E型（S46）来计算发送器和接收器之间的语音质量测量值。
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公开(公告)号：KR1020070059863A
公开(公告)日：2007-06-12
申请号：KR1020060050906
申请日：2006-06-07
CPC classification number: H04L65/80 , H04L43/0829 , H04L43/087 , H04M7/006
Abstract: A method and a system for measuring voice quality with QoS(Quality of Service) assurance in a VoIP(Voice over Internet Protocol) network are provided to offer voice quality having good QoS by calculating values related to parameters detrimental to voice quality when a voice quality measurement request is made, judging the fault state of a VoIP network connected between users, and removing fault factors. A system for measuring voice quality with QoS assurance comprises a VoIP network(380), a voice quality measurement server(500), and soft phone terminals(100,200). The VoIP network(380) provides a voice call service between transmitting and receiving units through packet transmission using an IP. The voice quality measurement server(500) creates and transmits a reference voice file for a voice quality measurement request received in the middle of telephone conversation between the transmitting and receiving units, packetizes the created reference voice file, converts it into test voice data, and transmits the converted data. Either of the soft phone terminals(100,200) transmits the voice quality measurement request to the voice quality measurement server(500) according to an inputted command, removes voice quality deterioration parameters, due to a fault of the VoIP network(380) generated according to distortion of the test voice data, on the basis of the reference voice file, and provides a voice quality service with QoS assurance.
Abstract translation: 提供了一种用于通过VoIP（语音电话协议）网络中的QoS（服务质量）保证来测量语音质量的方法和系统，以通过计算与语音质量有关的参数相关的值来提供具有良好QoS的语音质量，当语音质量 进行测量请求，判断用户间连接的VoIP网络的故障状态，消除故障因素。 用于通过QoS保证测量语音质量的系统包括VoIP网络（380），语音质量测量服务器（500）和软电话终端（100,200）。 VoIP网络（380）通过使用IP的分组传输在发送和接收单元之间提供语音呼叫服务。 语音质量测量服务器（500）创建并发送用于在发送和接收单元之间的电话会话中间接收的语音质量测量请求的参考语音文件，将创建的参考语音文件分组，将其转换为测试语音数据，以及 发送转换后的数据。 软电话终端（100,200）中的任一个根据输入的命令向语音质量测量服务器（500）发送语音质量测量请求，由于根据根据本发明的VoIP网络（380）的故障而导致的话音质量劣化参数的消除 基于参考语音文件对测试语音数据进行失真，并提供具有QoS保证的语音质量服务。
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公开(公告)号：KR1020080044743A
公开(公告)日：2008-05-21
申请号：KR1020070043248
申请日：2007-05-03
IPC: G10L21/0208 , H04L12/66
Abstract: A noise removal apparatus for a VoIP(Voice over Internet Protocol) service and a method therefor are provided to remove a noise signal in consideration of an SNR(Signal-to-Noise Ratio) weight value. A power spectrum estimator(110) estimates a voice power spectrum and a noise power spectrum from a noise voice signal. A Wiener filter designing unit(120) calculates an SNR from the voice power spectrum and the noise power spectrum, acquires an SNR weight value corresponding to the calculated SNR, and designs a Wiener filter function which reflects the SNR weight value. A Wiener filter unit(130) removes a noise signal included in the noise voice signal by using the Wiener filter function.
Abstract translation: 提供了一种用于VoIP（因特网协议语音）服务的噪声消除装置及其方法，以便考虑到SNR（信噪比）权重值来消除噪声信号。 功率谱估计器（110）从噪声语音信号估计语音功率谱和噪声功率谱。 维纳滤波器设计单元（120）根据语音功率谱和噪声功率谱计算SNR，获取对应于所计算的SNR的SNR权重值，并设计反映SNR权重值的维纳滤波函数。 维纳滤波器单元（130）通过使用维纳滤波器功能去除包含在噪声语音信号中的噪声信号。
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9.发明授权
公开(公告)号：KR100772199B1
公开(公告)日：2007-11-01
申请号：KR1020060096559
申请日：2006-09-29
Applicant: 한국전자통신연구원
CPC classification number: H04L65/80 , H04L65/1059
Abstract: A speech noise canceling apparatus and method for guaranteeing QoS(Quality of Service) of a VoIP service, and a VoIP(Voice over Internet Protocol) terminal employing the same are provided to measure noise generated between VoIP soft phones by using a PESQ(Perceptual Evaluation of Speech Quality) speech quality measurement algorithm and a voice data code book and cancel its noise. An input unit(122) receives a cancel signal of noise generated during voice call communication from the user. A file/data generating unit(142) generates a reference voice file corresponding to actual voice data in a VoIP network, packetizes the reference voice file, and converts the packetized reference voice file into test voice data. A PESQ measurement unit(140) extracts a PESQ-MOS(Mean Opinion Score) value by applying the reference voice file and the test voice data to a pre-set PESQ speech quality measurement algorithm. A parameter calculating unit(150) calculates a speech quality disincentive value corresponding to the generated noise by using the extracted PESQ-MOS value. A deviation value calculating unit(160) calculates the deviation between the calculated speech quality disincentive value and a pre-set speech quality disincentive value. A network management unit(170) determines an error state of the VoIP network based on the calculated deviation value and removes the speech quality disincentive.
Abstract translation: 提供一种用于保证VoIP服务的QoS（服务质量）和采用该语音服务的VoIP（语音网际协议）终端的语音消除装置和方法，以通过使用PESQ（感知评估）来测量VoIP软电话之间产生的噪声 的语音质量）语音质量测量算法和语音数据码本，并消除其噪声。 输入单元（122）从用户接收在语音呼叫通信期间产生的噪声消除信号。 文件/数据生成单元（142）生成与VoIP网络中的实际语音数据相对应的参考语音文件，分组参考语音文件，将分组化的参考语音文件转换为测试语音数据。 PESQ测量单元（140）通过将参考语音文件和测试语音数据应用于预设的PESQ语音质量测量算法来提取PESQ-MOS（Mean Opinion Score）值。 参数计算单元（150）通过使用所提取的PESQ-MOS值来计算与所生成的噪声相对应的语音质量抑制值。 偏差值计算单元（160）计算计算出的语音质量抑制值与预定语音质量抑制值之间的偏差。 网络管理单元（170）根据所计算的偏差值来确定VoIP网络的错误状态，并消除语音质量下降。
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公开(公告)号：KR1020100065045A
公开(公告)日：2010-06-15
申请号：KR1020090038476
申请日：2009-04-30
Abstract: PURPOSE: A method for measuring block distortion based on a reduced reference scheme is provided to support quick processing speed using a reduced reference scheme for measuring the quality of a received processing image. CONSTITUTION: An edge image of an original image is extracted. A block edge is extracted(411-414). A weighted value of the extracted block edge is determined. A block distortion value of the original image is calculated using the calculated weighted value(415). The received image edge image is extracted(422). The gradient size and direction of the edge image are calculated. A block edge is extracted(424).
Abstract translation: 目的：提供一种基于减少的参考方案来测量块失真的方法，以使用简化的参考方案来支持快速处理速度，用于测量接收到的处理图像的质量。 构成：提取原始图像的边缘图像。 提取块边缘（411-414）。 确定提取的块边缘的加权值。 使用计算的加权值来计算原始图像的块失真值（415）。 提取接收到的图像边缘图像（422）。 计算边缘图像的梯度大小和方向。 提取块边缘（424）。
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