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71.发明授权부가정보 비트스트림 변환을 포함하는 다양한 채널로구성된 다객체 오디오 신호의 부호화 및 복호화 장치 및방법 有权用于编码和解码具有各种信道的多对象音频信号的装置和方法，包括信息比特流转换
公开(公告)号：KR101086347B1
公开(公告)日：2011-11-23
申请号：KR1020070138847
申请日：2007-12-27
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
Abstract: 다양한 채널로 구성된 다객체 오디오 신호의 부호화 및 복호화 장치 및 방법에 관한 것으로서, 다양한 채널을 갖는 다객체 오디오 신호를 부호화 및 복호화하며, 종래SAC 비트스트림에 대한 역호환성을 제공하는 다양한 채널로 구성된 다객체 오디오 신호 부호화 및 복호화 장치 및 방법이 개시됨.
부호화 장치로 입력되는 오디오 객체 신호를 공간큐 기반으로 부호화하고 상기 부호화된 오디오 객체 신호에 대한 랜더링 정보를 생성하는 오디오 객체 부호화 수단을 포함하되, 상기 랜더링 정보는 상기 오디오 객체 신호에 대한 공간큐 정보; 상기 오디오 객체 신호의 채널 정보; 및 상기 오디오 객체 신호의 식별 정보를 포함하는 부호화 장치가 개시됨.
오디오 신호의 부호화 및 복호화에서 이용됨.
오디오 객체, 다객체, 비트스트림-
72.发明公开객체기반 오디오 파일 제공 장치 및 객체기반 오디오 파일 재생 장치, 그리고, 객체기반 오디오 파일 제공 방법 및 객체기반 오디오 재생 방법 有权提供和复制基于对象的音频文件的设备和方法
公开(公告)号：KR1020110033027A
公开(公告)日：2011-03-30
申请号：KR1020100082997
申请日：2010-08-26
Applicant: 한국전자통신연구원
CPC classification number: G11B20/10 , G11B20/12 , G11B2020/10546 , H04N21/439
Abstract: PURPOSE: An object-based audio file providing device, an object-based audio file reproducing device are provided to perform an object-based audio service even in a user terminal of a low specification. CONSTITUTION: An object based audio file includes a file header for an object-based audio service and a frame with a sound source which mixed each frame and total audio objects. An object-based audio file is received(S701). According to specification of an object-based audio file reproducing device, the object-based audio file is reproduced by controlling sound source which mixed the total audio objects(S702).
Abstract translation: 目的：提供一种基于对象的音频文件提供装置，即基于对象的音频文件再现装置，即使在低规格的用户终端中也执行基于对象的音频服务。 构成：基于对象的音频文件包括用于基于对象的音频服务的文件头和具有混合每个帧的音源的帧和总音频对象。 接收基于对象的音频文件（S701）。 根据基于对象的音频文件再现装置的规格，通过控制混合了总体音频对象的声源再现基于对象的音频文件（S702）。
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公开(公告)号：KR1020100114449A
公开(公告)日：2010-10-25
申请号：KR1020090113320
申请日：2009-11-23
Applicant: 한국전자통신연구원
Abstract: PURPOSE: An access system selection device in heterogeneous wireless communication system and a method thereof are provided to select an access system based on a plurality of attributes and to receives a service, thereby preventing a service quality conveniently. CONSTITUTION: An access system specifying module(205) connects to an access system by activation of a relevant module in an access system. A CPC(Cognitive Pilot Channel, Common Pilot Channel) specific module(210) receives information of each access system and frequency allocation information of each access system and a load level from a CPC server. If a QoS(Quality of Service) is under the predetermined threshold value, a control unit(220) selects an optimum access system.
Abstract translation: 目的：提供异构无线通信系统中的接入系统选择设备及其方法，用于基于多个属性选择接入系统并接收业务，从而方便地防止业务质量。 构成：访问系统指定模块（205）通过激活访问系统中的相关模块来连接到访问系统。 CPC（认知导频信道，公共导频信道）特定模块（210）从CPC服务器接收每个接入系统的信息和每个接入系统的频率分配信息和负载水平。 如果QoS（服务质量）低于预定阈值，则控制单元（220）选择最佳接入系统。
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公开(公告)号：KR1020100072476A
公开(公告)日：2010-07-01
申请号：KR1020080130896
申请日：2008-12-22
CPC classification number: G10L19/167 , G10L19/008 , H04S2400/11
Abstract: PURPOSE: A method and an apparatus for generating and playing an object based audio contents are provided to continuously play plural object-based audio contents according to the same audio preset. CONSTITUTION: A preset generating unit(110) generates an audio preset, and the audio preset includes the information on the plural audio objects which constitute audio contents. A preset ID setup unit(120) sets up the preset ID of the generated audio preset, and an encoding unit(140) encodes the plural audio objects, the audio preset and the preset ID. The encoding unit generates the object based audio contents.
Abstract translation: 目的：提供一种用于生成和播放基于对象的音频内容的方法和装置，以根据相同的音频预设来连续播放多个基于对象的音频内容。 构成：预设生成单元（110）生成音频预设，并且音频预设包括关于构成音频内容的多个音频对象的信息。 预设ID设定单元（120）设定所生成的音频预设的预设ID，编码单元（140）对多个音频对象，音频预设和预置ID进行编码。 编码单元生成基于对象的音频内容。
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公开(公告)号：KR1020100063639A
公开(公告)日：2010-06-11
申请号：KR1020090064918
申请日：2009-07-16
Applicant: 한국전자통신연구원
IPC: G10L19/008 , G11B20/10 , H04N21/439
Abstract: PURPOSE: A multichannel audio decoding apparatus based on the sound source localization proviso and a method thereof are provided, which compress the multi-channel audio signal, and then, transmit the stereo signal through the core stereo codec under compression condition, thereby the multichannel audio electrical transmission is realized. CONSTITUTION: An audio decoding part(220) restores the downmix signal to the former state. An integrated up-mixing unit(240) predicts the multi channel signal by the downmix signal and additional information bit stream. The integration up-mixing part creates the up-mixing signal with the up-mixing of the MIX signal. An integrated filter bank windowing unit(250) extracts the temporal area signal by the performance of the integrated filter bank in the up-mixing signal. The integrated filter bank windowing part extracts the output signal by the windowing of the up-mixing signal.
Abstract translation: 目的：提供一种基于声源定位条件的多声道音频解码装置及其方法，其压缩多声道音频信号，然后在压缩条件下通过核心立体声编解码器发送立体声信号，从而多声道音频 实现电气传输。 构成：音频解码部件（220）将下混合信号恢复到前一状态。 集成的上混合单元（240）通过下混信号和附加信息比特流来预测多声道信号。 集成上混合部分通过MIX信号的上混合创建上混合信号。 集成滤波器组开窗单元（250）通过上混合信号中的积分滤波器组的性能来提取时间区域信号。 积分滤波器组窗口部分通过加窗信号提取出输出信号。
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Patent Agency Ranking
公开(公告)号：KR1020100041678A
公开(公告)日：2010-04-22
申请号：KR1020090096888
申请日：2009-10-12
Applicant: 한국전자통신연구원
CPC classification number: G10L19/087 , G10L19/125 , G10L19/22 , G10L19/26
Abstract: PURPOSE: An encoding and decoding apparatus for a linear predictive coder residual signal of a modified discrete cosine transform based an unified speech and an audio coding are provided to perform encoding/decoding by expressing a residual signal as a complex signal. CONSTITUTION: A signal analyzing unit(110) selects a coding method of a signal filtered LPC. A first encoding unit(120) encodes a residual signal of LPC based on real filter bank according to the selection of the signal analyzing unit. A second encoding unit encodes a residual signal of LPC based on complex filter bank according to the selection of the signal analyzing unit. A third encoding unit encodes a residual signal of LPC based on ACELP(Algebraic Code Excited Linear Prediction) according to the selection of the signal analyzing unit.
Abstract translation: 目的：提供一种用于基于统一语音和音频编码的修正离散余弦变换的线性预测编码器残差信号的编码和解码装置，以通过将残差信号表示为复信号来执行编码/解码。 构成：信号分析单元（110）选择滤波后的LPC信号的编码方法。 第一编码单元（120）根据信号分析单元的选择，根据实数滤波器组对LPC的残留信号进行编码。 第二编码单元根据信号分析单元的选择，根据复数滤波器组对LPC的残差信号进行编码。 第三编码单元根据信号分析单元的选择，基于ACELP（代数码激励线性预测）对LPC的残留信号进行编码。
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公开(公告)号：KR1020100032843A
公开(公告)日：2010-03-26
申请号：KR1020090088524
申请日：2009-09-18
Applicant: 한국전자통신연구원 , 광운대학교 산학협력단
IPC: G10L19/18 , G10L19/022 , G10L19/12
CPC classification number: G10L19/0212 , G10L19/173 , G10L19/0204 , G10L19/18
Abstract: PURPOSE: An encoding and decoding device during conversion between a coder based on an MDCT and a heterogeneous coder are provided to integrate an audio coder based on an MDCT with another audio coder to minimize MDCT information when encoding or decoding an audio signal, thereby offsetting generated signal distortion. CONSTITUTION: When conversion between a coder based on an MDCT and a heterogeneous coder occurs, an encoding device(101) codes additional information to restore an input signal encoded according to a coding method based on an MDCT. The encoding device prevents production of an unnecessary bit stream. The encoding device encodes minimum additional information.
Abstract translation: 目的：提供在基于MDCT的编码器和异构编码器之间的转换期间的编码和解码装置，用于将基于MDCT的音频编码器与另一音频编码器集成，以在对音频信号进行编码或解码时最小化MDCT信息，从而抵消产生的 信号失真。 构成：当发生基于MDCT的编码器和异构编码器之间的转换时，编码装置（101）对附加信息进行编码以恢复根据基于MDCT的编码方法编码的输入信号。 编码设备防止不必要的比特流的产生。 编码装置编码最小附加信息。
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公开(公告)号：KR1020100007738A
公开(公告)日：2010-01-22
申请号：KR1020090061607
申请日：2009-07-07
Applicant: 한국전자통신연구원 , 광운대학교 산학협력단
CPC classification number: G10L19/20 , G10L19/0212 , G10L19/12
Abstract: PURPOSE: An apparatus for encoding and decoding integrated voice and audio signals is provided to combine a voice codec module with an audio codec module and selectively apply a codec module depending on a characteristic of an input signal, thereby showing excellent performance. CONSTITUTION: A module selecting unit(110) analyzes a characteristic of an input signal to select the first encoding module for encoding the first frame of an input signal. A voice encoding unit(130) encodes the input signal by selection of the module selecting unit to generate a voice bit string. An audio encoder(140) encodes the input signal by selection of the module selecting unit to generate an audio bit string. A bit stream generator(150) generates an output bit string from the audio encoder by selection of the module selecting unit.
Abstract translation: 目的：提供一种用于编码和解码集成语音和音频信号的装置，用于将语音编解码器模块与音频编解码器模块组合，并且根据输入信号的特性选择性地应用编解码器模块，从而显示出优异的性能。 构成：模块选择单元（110）分析输入信号的特性，以选择用于编码输入信号的第一帧的第一编码模块。 语音编码单元（130）通过选择模块选择单元对输入信号进行编码以产生语音比特串。 音频编码器（140）通过选择模块选择单元对输入信号进行编码以产生音频位串。 比特流生成器（150）通过选择模块选择单元从音频编码器生成输出比特串。
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公开(公告)号：KR100932790B1
公开(公告)日：2009-12-21
申请号：KR1020080036085
申请日：2008-04-18
Applicant: 한국전자통신연구원
IPC: G10L19/008 , G11B20/10 , H03M7/30
Abstract: 본 발명은 음원 간 상호상관을 이용한 멀티트랙 다운믹싱 장치 및 그 방법에 관한 것으로, 개별 음원 신호를 각 음원 간 상호상관을 이용하여 상호상관 값이 높은 조합을 가지는 멀티트랙 신호로 다운믹싱함으로써, 다운믹싱된 멀티트랙 신호로부터 개별 음원을 더욱 충실하게 복원할 수 있는, 음원 간 상호상관을 이용한 멀티트랙 다운믹싱 장치 및 그 방법을 제공하고자 한다.
이를 위하여, 본 발명은 멀티트랙 다운믹싱 장치에 있어서, 개별 음원 신호를 주파수 대역으로 변환하기 위한 신호 변환 수단; 상기 변환된 개별 음원 신호로부터 각 음원 간 공간정보를 산출하기 위한 공간정보 산출 수단; 상기 산출된 각 음원 간 공간정보를 이용하여 상호상관 값에 따라 믹싱조합 정보를 결정하기 위한 믹싱조합 결정 수단; 및 상기 결정된 믹싱조합 정보에 따라 상기 변환된 개별 음원 신호를 멀티트랙 신호로 다운믹싱하기 위한 멀티트랙 다운믹싱 수단을 포함한다.
상호상관, 멀티트랙, 개별 음원 신호, 공간정보, 믹싱조합, 폐루프, 믹싱 매트릭스 정보, 멀티트랙 다운믹싱Abstract translation: 本发明涉及一种使用该声源之间的互相关性的多轨道下混合设备和方法中，通过下混合使用互相关的各个声源信号到多轨道信号具有相应的声音源之间的高互相关值组合，向下 还有一种多声道缩混设备，利用声源之间的相互相关性，能够更真实地从混合多声道信号中恢复单个声源。
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80.发明授权
公开(公告)号：KR100931309B1
公开(公告)日：2009-12-11
申请号：KR1020070067101
申请日：2007-07-04
Applicant: 한국전자통신연구원
IPC: G10L19/008 , H03M7/30
CPC classification number: G10L19/008 , G10L19/02 , G10L19/18
Abstract: Provided is a method for controlling synchronizing downmix signals and MPEG surround side information signals by controlling a delay according to the kind of downmix audio signals in an MPEG surround decoder. When multi-channel audio signals are restored using an HE-AAC decoder and a low-power MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter. Anther delay unit delays spatial parameters to compensate for a delay caused in QMF and Nyquist banks when time-domain downmix signals are used. Also, when multi-channel audio signals are restored using an HE-AAC decoder and a high-quality MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter.
Abstract translation: 提供了一种用于通过根据MPEG环绕解码器中的下混音频信号的种类来控制延迟来控制同步下混信号和MPEG环绕旁侧信息信号的方法。 当使用HE-AAC解码器和低功率MPEG环绕解码器恢复多声道音频信号并且将从HE-AAC解码器输出的复杂QMF信号用作下混信号时，延迟单元补偿在实际 到复数转换器。 当使用时域下混合信号时，另一延迟单元延迟空间参数以补偿在QMF和奈奎斯特组中引起的延迟。 而且，当使用HE-AAC解码器和高质量MPEG环绕解码器来恢复多声道音频信号并且从HE-AAC解码器输出的复数QMF信号被用作下混音信号时，延迟单元补偿 一个真正的复杂转换器。
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