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131.发明公开
公开(公告)号：KR1020130054159A
公开(公告)日：2013-05-24
申请号：KR1020120127499
申请日：2012-11-12
Applicant: 한국전자통신연구원
IPC: G10L19/005
CPC classification number: G10L19/008 , G10L19/24
Abstract: PURPOSE: An encoding device for supporting a scalable multi-channel audio signal, a decoding device, and a method thereof are provided to restore and compress a multi-channel audio signal for providing three-dimensional audio in a realistic broadcasting environment. CONSTITUTION: A signal generator(201) generates a compatible multi-channel audio signal using a multi-channel audio signal and an audio object signal. An encoding unit(202) generates a first bit stream through a hierarchical encoding of the compatible multi-channel audio signal. A second encoding unit(203) generates a second bit stream through the encoding of the audio object signal. A bit stream formatter(204) generates an output bit stream using the first and second bit streams. [Reference numerals] (101) Encoding device; (201) Signal generator; (202) First encoding unit; (203) Second encoding unit; (204) Bit stream formatter; (AA) Multichannel audio signal; (BB) Audio object signal; (CC) Compatible multichannel audio signal; (DD) Second bit stream; (EE) First bit stream; (FF) First additional information; (GG) Second additional information; (HH) Third additional information; (II) Output bit stream
Abstract translation: 目的：提供一种用于支持可扩展多声道音频信号的编码装置，解码装置及其方法，用于恢复和压缩用于在逼真的广播环境中提供三维音频的多声道音频信号。 构成：信号发生器（201）使用多声道音频信号和音频对象信号产生兼容的多声道音频信号。 编码单元（202）通过兼容多声道音频信号的分层编码产生第一比特流。 第二编码单元（203）通过音频对象信号的编码产生第二位流。 比特流格式化器（204）使用第一和第二比特流产生输出比特流。 （101）编码装置; （201）信号发生器; （202）第一编码单元; （203）第二编码单元; （204）比特流格式化器; （AA）多声道音频信号; （BB）音频对象信号; （CC）兼容的多声道音频信号; （DD）第二位流; （EE）第一位流; （FF）第一附加信息; （GG）第二附加信息; （HH）第三次附加信息; （II）输出比特流
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公开(公告)号：KR101159888B1
公开(公告)日：2012-06-25
申请号：KR1020080127927
申请日：2008-12-16
Applicant: 한국전자통신연구원
Abstract: 본 발명의 일 실시예에 따른 다중 액세스 가능한 무선/이동 시스템 환경에서의 AS 탐색 방법은 서버로부터 수신되는 배치 정보를 이용하여, 이동단말에 서비스를 제공하는 제1 셀의 AS(Access System)와, 상기 제1 셀과 인접한 제2 셀의 AS 사이의 제1 거리를 산출하는 단계; 상기 제1 셀의 AS로부터 수신되는 수신신호의 신호세기를 측정하여, 상기 이동단말과 상기 제1 셀의 AS 사이의 제2 거리를 산출하는 단계; 상기 제1 거리 및 상기 제2 거리를 이용하여 상기 이동단말의 위치를 추정하는 단계; 및 상기 추정된 위치를 이용하여 상기 제2 셀의 AS 탐색에 관련한 인터페이스의 활성화 주기를 조절하는 단계를 포함한다.
다중 액세스, 액세스 시스템(AS), AS 탐색, 셀 배치 정보, 수신신호강도(RSS)-
133.发明公开
公开(公告)号：KR1020120031998A
公开(公告)日：2012-04-04
申请号：KR1020120030545
申请日：2012-03-26
Applicant: 한국전자통신연구원
IPC: G10L19/00
CPC classification number: G10L19/008 , G10L19/02 , G10L19/18
Abstract: PURPOSE: A device for restoring multichannel audio signal using an HE-AAC(High-Efficiency Advanced Audio Coding) decoder and an MPEG surround decoder and a method thereof are provided to add a delay stage when connecting a downmix signal of the HE-AAC decoder to a QMF(Quadrature Mirror Filter) signal of the real number domain, a QMF signal of the complex number domain, and a time domain signal. CONSTITUTION: When a downmix signal of a time domain encoded by HE-AAC is combined with an MPEG surround low power decoder, a delay stage(501) applies delay to extracted spatial parameters. The delay is identical to the sum of the delay of HE-AAC QMF synthetic filter process, real QMF(502) and Nyquist analysis filter(504) process of the MPEG surround low power decoder, and a real to complex converter(503).
Abstract translation: 目的：提供一种用于使用HE-AAC（高效率高级音频编码）解码器和MPEG环绕解码器及其方法恢复多声道音频信号的装置，用于在连接HE-AAC解码器的下混合信号时增加延迟级 到实数域的QMF（正交镜像滤波器）信号，复数域的QMF信号和时域信号。 构成：当由HE-AAC编码的时域的下混合信号与MPEG环绕低功率解码器组合时，延迟级（501）对提取的空间参数应用延迟。 该延迟与MPEG环绕低功率解码器的HE-AAC QMF合成滤波处理，实际QMF（502）和奈奎斯特分析滤波器（504）处理的延迟和真实到复杂转换器（503）的总和相同。
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公开(公告)号：KR1020110036023A
公开(公告)日：2011-04-06
申请号：KR1020110014763
申请日：2011-02-18
Applicant: 한국전자통신연구원
IPC: G10L19/008 , G10L19/20 , H03M7/30 , H04N21/233
CPC classification number: G10L19/008 , G10L19/0017 , G10L19/173 , H04S3/002 , H04S7/30 , H04S2400/11 , H04S2420/03 , G10L19/00
Abstract: PURPOSE: An apparatus and method for coding and decoding multi-object audio signal are provided to perform coding and decoding multi-object audio signal which is composed of various channels providing reverse compatibility about the existing SAC bitstream. CONSTITUTION: An audio object coding unit codes audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals. The rendering information includes spatial cue information for audio-object signals, channel information of the audio-object signals, and identification information of the audio-object signals.
Abstract translation: 目的：提供一种用于对多对象音频信号进行编码和解码的装置和方法，用于对由现有SAC比特流提供反向兼容性的各种信道组成的多对象音频信号进行编码和解码。 构成：音频对象编码单元基于空间提示对输入到编码装置的音频对象信号进行编码，并生成编码音频对象信号的绘制信息。 呈现信息包括用于音频对象信号的空间提示信息，音频对象信号的频道信息以及音频对象信号的识别信息。
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公开(公告)号：KR1020110029055A
公开(公告)日：2011-03-22
申请号：KR1020090122218
申请日：2009-12-10
Applicant: 한국전자통신연구원
IPC: G10L21/0272 , G10L25/81
Abstract: PURPOSE: A method and system for separating a music sound source without using a sound source database are provided to separate a sound source generated using a rhythm musical instrument based on characteristics of the rhythm musical instrument repeated in an aspect of time, and thereby may separate a sound source included in a mixed signal even when a learning database generated using a specific sound source is absent. CONSTITUTION: An NMPCF(Nonnegative Matrix Partial Co-Factorization) analysis unit(130) obtains info nation shared by the plurality of segments in accordance with an NMPCF algorithm. A rhythm instrument signal has frequency characteristics such as a pitch, that may not be easily changed, and may be repeatedly generated, whereby the shared information may correspond to information of a rhythm musical instrument. A target instrument signal separating unit(140) separates a target instrument signal corresponding to a specific sound source from the mixed signal by calculating an inner product between the entity matrices obtained by the NMPCF analysis unit. The target instrument signal is a signal including sounds generated using the rhythm musical instrument. A signal association unit(150) associates the target instrument signals for each of the plurality of segments separated in the target instrument signal separating unit.
Abstract translation: 目的：提供一种用于在不使用声源数据库的情况下分离音乐声源的方法和系统，用于基于在一段时间内重复的节奏乐器的特征来分离使用节奏乐器产生的声源，从而可以分离 即使当使用特定声源生成的学习数据库不存在时，也包含在混合信号中的声源。 构成：NMPCF（非负矩阵部分协同因子分解）分析单元（130）根据NMPCF算法获得由多个段共享的信息国。 节奏乐器信号具有诸如间距的频率特性，其可能不容易改变，并且可以重复地产生，由此共享信息可以对应于节奏乐器的信息。 目标仪器信号分离单元（140）通过计算由NMPCF分析单元获得的实体矩阵之间的内积，将与特定声源相对应的目标仪器信号与混合信号分离。 目标仪器信号是包括使用节奏乐器产生的声音的信号。 信号关联单元（150）将目标仪器信号分离单元中分离出的多个分段中的每个分段的目标仪器信号相关联。
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Patent Agency Ranking136.发明公开객체기반 오디오 서비스를 위한 비트스트림 구조화 방법, 비트스트림 구조화 장치 및 비트스트림 부호화 장치 有权用于基于对象的音频服务的BIT-STREAM结构的装置和方法以及用于编码位流的装置
公开(公告)号：KR1020110023693A
公开(公告)日：2011-03-08
申请号：KR1020090127946
申请日：2009-12-21
Applicant: 한국전자통신연구원
IPC: G10L19/20 , H04N5/60 , H04N21/233 , G11B20/10
Abstract: PURPOSE: An apparatus and a method for structuring bitstream for an object-based audio service are provided to reduce a degradation in the sound quality occurring due to an excessive volume control by designating an upper bound value and a lower bound value of a reproduction volume in an object-based audio service. CONSTITUTION: A bitstream splitting unit divides a bitstream into a file header, and an audio object frame through sound source splitting(110). A reproduction level information storage unit stores the reconstruction level information for the regeneration of the audio object within the file header(120). A preset storage unit stores the preset information for the regeneration of the audio object within the file header(130).
Abstract translation: 目的：提供一种用于构建用于基于对象的音频服务的比特流的装置和方法，以通过指定再现音量的上限值和下限值来减少由于过度音量控制而产生的声音质量下降 一种基于对象的音频服务。 构成：比特流分割单元通过声源分割将比特流分成文件头和音频对象帧（110）。 再现级别信息存储单元将用于音频对象再生的重建级别信息存储在文件头部（120）内。 预设存储单元将用于音频对象的再生的预设信息存储在文件头部（130）内。
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137.发明授权
公开(公告)号：KR100971700B1
公开(公告)日：2010-07-22
申请号：KR1020080043578
申请日：2008-05-09
Applicant: 한국전자통신연구원
IPC: G10L19/008 , H03M7/30 , H04S5/00
Abstract: 본 발명은 공간큐 기반의 바이노럴 스테레오 합성 장치 및 그 방법과, 그를 이용한 바이노럴 스테레오 복호화 장치에 관한 것으로, 공간큐 정보(가상음원 위치정보)와 각 채널별 바이노럴 필터 계수에 기초하여 좌/우 오디오 신호를 주파수 영역에서 필터링하고, 그 필터링된 신호를 바이노럴 스테레오 신호로 복호화함으로써, 주파수 영역에서의 선형 복적분을 통해 고품질의 바이노럴 스테레오 신호를 용이하게 재생할 수 있게 하는, 공간큐 기반의 바이노럴 스테레오 합성 장치 및 그 방법과, 그를 이용한 바이노럴 스테레오 복호화 장치를 제공하고자 한다.
이를 위하여, 본 발명은, 공간큐 기반의 바이노럴 스테레오 합성 장치에 있어서, 가상음원 위치 기반의 공간큐 정보를 합성하여 서브밴드별 각 채널의 파워 이득값을 산출하기 위한 공간큐 정보 합성 수단; 상기 산출된 각 채널의 파워 이득값과 바이노럴 필터 계수에 기초하여, 좌/우 오디오 신호를 주파수 영역에서 필터링하기 위한 신호 필터링 수단; 및 상기 필터링된 신호를 시간 영역으로 역변환하기 위한 신호 역변환 수단을 포함한다.
공간큐 정보, 가상음원 위치정보, 바이노럴 필터 계수, 바이노럴 스테레오 합성, 바이노럴 스테레오 복호화, 공간큐 기반의 오디오 코딩, 3차원 오디오-
公开(公告)号：KR1020100069278A
公开(公告)日：2010-06-24
申请号：KR1020080127927
申请日：2008-12-16
Applicant: 한국전자통신연구원
Abstract: PURPOSE: A method and an apparatus for discovering an access system in multi-accessible wireless/mobile system environment are provided to efficiently search an AS by using the position of a mobile terminal and reception power strength. CONSTITUTION: A calculation unit(110) calculates a first distance between an AS of a first cell and an AS of the second cell, wherein the first distance is the range of providing a service to a mobile terminal. The calculation unit calculates the distance between the mobile terminal and the AS of the first cell, and measures the signal strength of the reception signal received from the AS of the first cell. An estimation unit(120) estimates the position of the mobile terminal by using the first and second distances. A control unit(130) controls the activation period of an interface related to the AS search of the second cell by using the estimated position.
Abstract translation: 目的：提供一种用于在多方面的无线/移动系统环境中发现接入系统的方法和装置，以通过使用移动终端的位置和接收功率强度来有效地搜索AS。 构成：计算单元（110）计算第一小区的AS与第二小区的AS之间的第一距离，其中第一距离是向移动终端提供服务的范围。 计算单元计算移动终端与第一小区的AS之间的距离，并测量从第一小区的AS接收的接收信号的信号强度。 估计单元（120）通过使用第一和第二距离来估计移动终端的位置。 控制单元（130）通过使用估计位置来控制与第二小区的AS搜索相关的接口的激活周期。
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公开(公告)号：KR1020100059726A
公开(公告)日：2010-06-04
申请号：KR1020090114783
申请日：2009-11-25
Applicant: 한국전자통신연구원 , 광운대학교 산학협력단
IPC: G10L19/04 , G10L19/18 , H03M7/30 , H04N21/233
CPC classification number: G10L19/22 , G10L19/022 , G10L19/20
Abstract: PURPOSE: A unified speech/audio coder(USAC) processing windows sequence based mode switching is provided to perform encoding or decoding by processing different window sequences for different situation, thereby improving coding efficiency. CONSTITUTION: A mode switching unit performs inter-linear prediction domain mode switching for the sub-frames which constitute a frame of an input signal. An encoding unit applies a window sequence to current encoding subframes, wherein the window sequence is based on a switched LPD(Linear Prediction Domain) mode. In addition, the encoding unit applies a changed window sequence to the current encoding subframe according to the LPD modes of previous and following subframes.
Abstract translation: 目的：提供统一的语音/音频编码器（USAC）处理基于Windows序列的模式切换，通过对不同情况处理不同的窗口序列进行编码或解码，从而提高编码效率。 构成：模式切换单元对构成输入信号的帧的子帧进行线间预测域模式切换。 编码单元将窗口序列应用于当前编码子帧，其中窗口序列基于切换的LPD（线性预测域）模式。 此外，编码单元根据前一个和后一个子帧的LPD模式将改变的窗口序列应用于当前编码子帧。
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公开(公告)号：KR1020100020889A
公开(公告)日：2010-02-23
申请号：KR1020080130999
申请日：2008-12-22
Applicant: 한국전자통신연구원 , 티유미디어 주식회사 , 주식회사 이머시스
IPC: H04N21/233
CPC classification number: H04N21/2335 , H04H2201/11
Abstract: PURPOSE: A method and an apparatus for encoding and decoding audio signal are provided to offer the sound with sense of realism by supporting the play of the multi-channel audio signal. CONSTITUTION: A receiver(110) receives one signal among the multi-channel audio signal and the first audio signal. A first encoding unit(120) extracts spatial parameter from the multi-channel audio signal, mixes the multi-channel audio signal, and generates the second audio signal. A secondary symbol stoker(130) encodes at least one of the first and the second audio signal, and generates the encoded audio signal. A multiplexer(140) multiplexes the encoded audio signal as described above and spatial parameter to generate the satellite DMB(Digital Multimedia Broadcasting) stream.
Abstract translation: 目的：提供一种用于对音频信号进行编码和解码的方法和装置，以通过支持多声道音频信号的播放来提供具有现实感的声音。 构成：接收机（110）在多声道音频信号和第一音频信号之间接收一个信号。 第一编码单元（120）从多声道音频信号中提取空间参数，混合多声道音频信号，并产生第二音频信号。 辅助符号加载器（130）对第一和第二音频信号中的至少一个进行编码，并产生编码的音频信号。 多路复用器（140）如上所述复用经编码的音频信号和空间参数，以产生卫星DMB（数字多媒体广播）流。
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