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    • 1. 发明授权
    • Method and apparatus for matching sound quality measurement sections of variable bandwidth multi-codec
    • 用于匹配可变带宽多编解码器的音质测量部分的方法和装置
    • US08117029B2
    • 2012-02-14
    • US11928557
    • 2007-10-30
    • Dae-Ho KimTae-Gyu KangKi-Jong KooDo Young KimHae Won Jung
    • Dae-Ho KimTae-Gyu KangKi-Jong KooDo Young KimHae Won Jung
    • G10L21/00
    • G10L25/69G10L19/18
    • Provided is a transmission apparatus for matching sound quality measurement sections of a variable bandwidth multi-codec. The apparatus includes a measurement section setting unit setting a measurement section, which is to be measured for sound quality, in units of time; a first conversion unit converting the measurement section into a measurement section in units of samples; and an information synthesis unit synthesizing information regarding the measurement section in units of samples with a digital original sound and outputting the synthesis result. In addition, provided is a method of matching a measurement section of a reference sound, based on which the end-to-end sound quality measurement of the variable bandwidth multi-codec is performed, and a measurement section of a sound produced by the variable bandwidth multi-codec in a real-time Internet multimedia service. Therefore, distortion of measurement results due to un-matching measurement sections can be reduced.
    • 提供了一种用于匹配可变带宽多编解码器的音质测量部分的发送装置。 该装置包括以时间为单位设置要测量声音质量的测量部分的测量部分设置单元; 第一转换单元,以样本为单位将测量部转换为测量部; 以及信息合成单元,以具有数字原始声音的样本为单位合成关于测量部分的信息,并输出合成结果。 此外,提供了一种将参考声音的测量部分与可变带宽多编解码器的端对端音质测量进行匹配的方法以及由变量产生的声音的测量部分 带宽多编解码器在实时互联网多媒体服务。 因此,可以减少由于不匹配的测量部分引起的测量结果的失真。
    • 2. 发明授权
    • Apparatus and method for measuring quality of sound encoded with a variable band multi-codec
    • 用可变频带多编解码器编码的声音质量的装置和方法
    • US07986634B2
    • 2011-07-26
    • US11857539
    • 2007-09-19
    • Tae-Gyu KangKi-Jong KooDae-Ho KimDo Young KimHae Won Jung
    • Tae-Gyu KangKi-Jong KooDae-Ho KimDo Young KimHae Won Jung
    • H04L12/26
    • G10L25/69G10L19/24
    • Provided are a method and apparatus for measuring sound quality in a variable band multi-codec. The sound quality measurement apparatus includes: a recording file receiving/generating unit receiving a first recording file in which a natural sound is recorded, and a second recording file obtained by converting the natural sound into digital data using the variable band multi-codec, receiving information obtained by encoding the natural sound using the variable band multi-codec, in the format of a Real Time Protocol (RTP) packet, unpacking the RTP packet, decoding the RTP packet using the variable band multi-codec, and generating a third recording file; a Mean Opinion Score (MOS) value calculating unit repeatedly selecting a file from among the first recording file, the second recording file, and the third recording file, or selecting two files from among the first recording file, the second recording file, and the third recording file, and calculating a MOS value by obtaining a difference between the selected results; and a MOS value comparison unit comparing a plurality of MOS values generated by the MOS value calculating unit, with each other, and detecting a cause of sound quality deterioration.
    • 提供了一种用于测量可变频带多编解码器中的声音质量的方法和装置。 声音测量装置包括:记录文件接收/生​​成单元,接收其中记录有自然声音的第一记录文件;以及第二记录文件,其通过使用可变频带多编解码器将自然声音转换为数字数据而获得,第二记录文件接收 使用可变频带多编解码器以实时协议(RTP)分组的格式编码自然声音获得的信息,解包RTP分组,使用可变频带多编解码器解码RTP分组,以及生成第三记录 文件; 平均意见得分(MOS)值计算单元,重复从第一记录文件,第二记录文件和第三记录文件中选择文件,或者从第一记录文件,第二记录文件和第二记录文件中选择两个文件 第三记录文件,并且通过获得所选择的结果之间的差来计算MOS值; 以及MOS值比较单元,将由MOS值计算单元生成的多个MOS值彼此进行比较,并且检测出音质劣化的原因。
    • 4. 发明申请
    • Apparatus and method for minimizing number of transcodings in multi-network multi-codec environment
    • 用于最小化多网络多编解码器环境中的转码数量的装置和方法
    • US20080081648A1
    • 2008-04-03
    • US11702275
    • 2007-02-05
    • Tae Gyu KangKi Jong KooDae Ho KimDo Young KimHae Won Jung
    • Tae Gyu KangKi Jong KooDae Ho KimDo Young KimHae Won Jung
    • H04B7/00
    • H04M7/0072H04L65/1069H04L65/80
    • Provided is a method and apparatus for minimizing the number of transcodings between network devices in a multi-network multi-codec environment. The method includes: creating a received codec list by receiving a transmit codec comprised in a call setting message from a transmission device and the number of transcodings of the transmit codec, which has been performed from a codec of an initial transmission device; creating a codec Quality of Service (QoS) list containing total codecs of the multi-network and quality information of each of the total codecs; creating a transcodec list containing internally providable transcodecs and quality information of the transcodecs based on the codec QoS list; and creating an updated codec list by adding codecs of the transcodec list matching codecs of the received codec list to the received codec list and adjusting codec priority according to the number of transcodings. Accordingly, since the number of transcodings can be minimized, a quality decrease of original media can be minimized.
    • 提供了一种用于最小化多网络多编解码器环境中的网络设备之间的转码次数的方法和装置。 该方法包括:通过从传输设备接收包括在呼叫设置消息中的传输编解码器和已经从初始传输设备的编解码器执行的传输编解码器的转码次数来创建接收的编解码器列表; 创建包含多网络的总编解码器和每个编解码器的质量信息的编解码器服务质量(QoS)列表; 基于编解码器QoS列表创建包含内容可提供的转录代码和转录代码的质量信息的转录列表; 以及通过将接收到的编解码器列表的编解码器匹配的转码器列表的编解码器添加到接收到的编解码器列表并根据转码数量来调整编解码器优先级来创建更新的编解码器列表。 因此,由于能够最小化转码次数,因此可以使原始媒体的质量下降最小化。
    • 6. 发明申请
    • Apparatus and method of variable bandwidth multi-codec QoS control
    • 可变带宽多编解码器QoS控制的装置和方法
    • US20070133441A1
    • 2007-06-14
    • US11634561
    • 2006-12-06
    • Tae Gyu KangDo Young KimHae Won Jung
    • Tae Gyu KangDo Young KimHae Won Jung
    • H04B1/44
    • H04W28/18
    • An apparatus and method of variable bandwidth multi-codec quality of service (QoS) control are provided. The apparatus for controlling the QoS of a variable bandwidth multi-codec includes: a network state detection unit detecting a network state including at least one of a packet loss ratio, a packet loss interval, and a packet delay time based on an RTP packet transmitted to and received from a destination for which a call connection is established; and a codec control unit updating a transmission rate by comparing the detected resultant value with an already set reference value and increasing or decreasing the transmission rate, and controlling the variable bandwidth multi-codec to code data with the updated transmission rate. According to the apparatus and method, data can be coded with a codec transmission rate suitable for a network state identified during a voice call after the call is set up.
    • 提供了一种可变带宽多编解码器服务质量(QoS)控制的装置和方法。 用于控制可变带宽多编解码器的QoS的装置包括:网络状态检测单元,其基于发送的RTP分组检测包括分组丢失率,分组丢失间隔和分组延迟时间中的至少一个的网络状态 从建立呼叫连接的目的地接收和接收; 以及编解码器控制单元,通过将检测到的结果值与已经设定的参考值进行比较来增加或减少传输速率来更新传输速率,并且控制可变带宽多编解码器以更新的传输速率对数据进行编码。 根据该装置和方法,可以在呼叫建立之后利用适合于在语音呼叫期间识别的网络状态的编解码传输速率对数据进行编码。
    • 7. 发明授权
    • Apparatus and method of variable bandwidth multi-codec QoS control
    • 可变带宽多编解码器QoS控制的装置和方法
    • US07778177B2
    • 2010-08-17
    • US11634561
    • 2006-12-06
    • Tae Gyu KangDo Young KimHae Won Jung
    • Tae Gyu KangDo Young KimHae Won Jung
    • H04J1/16
    • H04W28/18
    • An apparatus and method of variable bandwidth multi-codec quality of service (QoS) control are provided. The apparatus for controlling the QoS of a variable bandwidth multi-codec includes: a network state detection unit detecting a network state including at least one of a packet loss ratio, a packet loss interval, and a packet delay time based on an RTP packet transmitted to and received from a destination for which a call connection is established; and a codec control unit updating a transmission rate by comparing the detected resultant value with an already set reference value and increasing or decreasing the transmission rate, and controlling the variable bandwidth multi-codec to code data with the updated transmission rate. According to the apparatus and method, data can be coded with a codec transmission rate suitable for a network state identified during a voice call after the call is set up.
    • 提供了一种可变带宽多编解码器服务质量(QoS)控制的装置和方法。 用于控制可变带宽多编解码器的QoS的装置包括:网络状态检测单元,其基于发送的RTP分组检测包括分组丢失率,分组丢失间隔和分组延迟时间中的至少一个的网络状态 从建立呼叫连接的目的地接收和接收; 以及编解码器控制单元,通过将检测到的结果值与已经设定的基准值进行比较,并且增加或减少传输速率来更新传输速率,并且控制可变带宽多编解码器以更新的传输速率对数据进行编码。 根据该装置和方法,可以在呼叫建立之后利用适合于在语音呼叫期间识别的网络状态的编解码传输速率对数据进行编码。
    • 8. 发明申请
    • WIRE AND WIRELESS INTERNET PHONE TERMINAL USING WIDEBAND VOICE CODEC
    • 使用宽带语音编解码器的无线和无线互联网终端
    • US20080212568A1
    • 2008-09-04
    • US11948956
    • 2007-11-30
    • Woon Seob SoDo Young KimHae Won Jung
    • Woon Seob SoDo Young KimHae Won Jung
    • H04L12/66
    • H04L12/66
    • A wire/wireless Internet phone terminal using a wideband voice codec is provided. The wire/wireless Internet phone terminal using a wideband voice codec includes: a multimedia application processor for including a process core to perform a protocol according to a wire and wireless interface communication scheme and supporting wideband voice service; an Ethernet processing unit for connecting the multimedia application processor to the Ethernet to perform an Ethernet physical-layer function of and transforming a power input from the Ethernet to supply a driving power to the multimedia application processor; a PSTN (public switched telephone network) processing unit for connected to the multimedia application processor and a PSTN to emulate a telephone function; and a wireless processing unit for connecting the multimedia application processor to an AP (access point) in a wireless manner. Accordingly, it is possible to provide a wideband service without a limitation to wire/wireless implementation.
    • 提供使用宽带语音编解码器的有线/无线互联网电话终端。 使用宽带语音编解码器的有线/无线因特网电话终端包括:多媒体应用处理器,用于包括处理核,以根据有线和无线接口通信方案执行协议并支持宽带语音服务; 以太网处理单元,用于将多媒体应用处理器连接到以太网,以执行以太网物理层功能并将来自以太网的电力输入变换为向多媒体应用处理器提供驱动电力; 用于连接到多媒体应用处理器的PSTN(公共交换电话网络)处理单元和用于模拟电话功能的PSTN; 以及用于以无线方式将多媒体应用处理器连接到AP(接入点)的无线处理单元。 因此,可以提供宽带业务,而不限于有线/无线实现。
    • 9. 发明申请
    • SPEECH CODING APPARATUS AND METHOD
    • 语音编码设备和方法
    • US20080140393A1
    • 2008-06-12
    • US11929922
    • 2007-10-30
    • Hyun-Woo KIMDo Young KimHae Won Jung
    • Hyun-Woo KIMDo Young KimHae Won Jung
    • G10L11/04
    • G10L19/0204G10L19/12G10L21/038G10L25/18
    • Provided is a speech coding apparatus and method. A band divider divides an input signal into a high-band signal and a low-band signal, a narrowband encoder encodes the low-band signal using a Code Excited Linear Prediction (CELP)-based narrowband speech codec, a frequency characteristic collector converts the high-band signal to a signal in a frequency domain and obtains Modified Discrete Cosine Transform (MDCT) coefficients, a subband determiner determines subbands in a final stage based on the MDCT coefficients and determines subbands for quantization based on the subbands in a final stage, a gain quantizer performs gain quantization of the subbands, a bit assignment unit assigns bits to the subbands according to the magnitude of the gain quantization, and a shape quantizer performs shape quantization of the subbands in an algebraic method. Accordingly, algorithm consistency can be maintained and a complexity can be reduced by extending a bandwidth with a small number of bits in a speech codec.
    • 提供了一种语音编码装置和方法。 频带分频器将输入信号分成高频带信号和低频带信号,窄带编码器使用基于码激励线性预测(CELP)的窄带语音编解码器对低频信号进行编码,频率特征收集器将 高频信号到频域中的信号并获得修正离散余弦变换(MDCT)系数,子带确定器基于MDCT系数确定最后一级中的子带,并且基于最终级中的子带确定用于量化的子带, 增益量化器执行子带的增益量化,比特分配单元根据增益量化的大小向子带分配比特,并且形状量化器以代数方法执行子带的形状量化。 因此,可以保持算法一致性,并且通过在语音编解码器中扩展带有少量位的带宽来降低复杂度。