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公开(公告)号：KR101315617B1
公开(公告)日：2013-10-08
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: 모드 스위칭에 기초하여 윈도우 시퀀스를 처리하는 통합 음성/오디오 부/복호화기가 개시된다. 통합 음성/오디오 부/복호화기는 모드 스위칭이 발생하는 경우, 폴딩 포인트를 기준으로 프레임 간 오버랩을 수행하여 부호화하거나 또는 복호화할 수 있다. 통합 음성/오디오 부/복호화기는 부호화 또는 복호화를 수행하기 위해 상황에 따라 다른 윈도우 시퀀스를 처리함으로써, 코딩 성능을 향상시킬 수 있다.
USAC(Unified Speech/Audio Coder), 윈도우 시퀀스, FD, MDCT, LPD-
公开(公告)号：KR1020130054976A
公开(公告)日：2013-05-27
申请号：KR1020130048838
申请日：2013-04-30
Applicant: 한국전자통신연구원 , 광운대학교 산학협력단
CPC classification number: G10L19/22 , G10L19/022 , G10L19/20
Abstract: PURPOSE: An integrated voice/audio coding/decoding device for processing a window sequence based on mode switching is provided to differently perform a coding method corresponding to a property of an input signal, thereby maximizing coding performance and sound quality. CONSTITUTION: An integrated voice/audio coding/decoding device performs LPD(Linear Prediction Domain) mode switching for a sub-frame which comprises a frame of an input signal. The integrated voice/audio coding/decoding device codes the input signal by applying a window based on an LPD mode to a current sub-frame. The integrated voice/audio coding/decoding device applies the window to the current sub-frame. The window is changed corresponding to an LPD mode of a previous sub-frame and an LPD mode of a next sub-frame.
Abstract translation: 目的：提供一种用于基于模式切换处理窗口序列的集成语音/音频编码/解码设备，以不同地执行与输入信号的属性相对应的编码方法，从而最大化编码性能和声音质量。 构成：集成语音/音频编码/解码装置对包括输入信号的帧的子帧执行LPD（线性预测域）模式切换。 集成语音/音频编码/解码装置通过将基于LPD模式的窗口应用于当前子帧来对输入信号进行编码。 集成语音/音频编码/解码装置将窗口应用于当前子帧。 该窗口对应于先前子帧的LPD模式和下一个子帧的LPD模式而改变。
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公开(公告)号：KR101261677B1
公开(公告)日：2013-05-06
申请号：KR1020090062070
申请日：2009-07-08
Applicant: 한국전자통신연구원 , 광운대학교 산학협력단
IPC: G10L19/04 , G10L19/18 , G10L19/008
CPC classification number: G10L19/12 , G10L19/008 , G10L19/02 , G10L19/20 , G10L19/22 , G11C2207/16
Abstract: 음성/음악통합신호의부호화/복호화장치가개시된다. 음성/음악통합신호의부호화장치는, 입력신호의특성을분석하는입력신호분석부, 상기입력신호가음악특성신호인경우, 상기입력신호를주파수영역으로변환하여부호화하는제1 변환부호화부, 상기입력신호가음성특성신호인경우, 상기입력신호를 LPCLinear Predictive Coding)기반부호화하는 LPC 부호화부및 상기제1 변환부호화부의출력신호및 상기 LPC 부호화부의출력신호를이용하여비트스트림을생성하는비트스트림생성부를포함한다.
Abstract translation: 提供了一种用于对语音信号和音频信号进行整体编码和解码的装置和方法。 编码装置可以包括：输入信号分析器，用于分析输入信号的特性; 第一转换编码器，用于将输入信号转换为频域信号，并且在输入信号是音频特征信号时对输入信号进行编码; 线性预测编码（LPC）编码器，用于当输入信号是语音特征信号时执行输入信号的LPC编码; 以及比特流发生器，用于使用第一转换编码器的输出信号和LPC编码器的输出信号来产生比特流。
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公开(公告)号：KR1020120089221A
公开(公告)日：2012-08-09
申请号：KR1020120076634
申请日：2012-07-13
Applicant: 한국전자통신연구원 , 광운대학교 산학협력단
IPC: G10L19/00
CPC classification number: G10L19/12 , G10L19/008 , G10L19/02 , G10L19/20 , G10L19/22 , G11C2207/16 , G10L19/00
Abstract: PURPOSE: A method for encoding/decoding an integrated voice/music signal and a device thereof are provided to effectively select an internal module according to a feature of an input signal, thereby providing superior sound quality for both a voice signal and a music signal at various bit rates. CONSTITUTION: An input signal analyzing unit(110) analyzes a feature of an input signal. When the input signal is a music feature signal, a first conversion encoding unit(120) converts a domain of the input signal into a frequency domain. The first conversion encoding unit encodes the converted input signal. When the input signal is a voice feature signal, an LPC(Linear Predictive Coding) encoding unit(130) encodes the input signal based on LPC. A bit stream generating unit(140) generates a bit stream using output signals of the first conversion encoding unit and the LPC encoding unit.
Abstract translation: 目的：提供一种用于对集成语音/音乐信号进行编码/解码的方法及其装置，以根据输入信号的特征有效地选择内部模块，从而为语音信号和音乐信号两者提供卓越的音质 各种比特率。 构成：输入信号分析单元（110）分析输入信号的特征。 当输入信号是音乐特征信号时，第一转换编码单元（120）将输入信号的域转换成频域。 第一转换编码单元对转换的输入信号进行编码。 当输入信号是语音特征信号时，LPC（线性预测编码）编码单元130根据LPC对输入信号进行编码。 比特流生成单元（140）使用第一转换编码单元和LPC编码单元的输出信号生成比特流。
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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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Patent Agency Ranking
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：KR100903110B1
公开(公告)日：2009-06-16
申请号：KR1020070036585
申请日：2007-04-13
Applicant: 한국전자통신연구원 , 한양대학교 산학협력단 , 광운대학교 산학협력단 , 연세대학교 산학협력단
Abstract: 본 발명은 트렐리스 부호 양자화 알고리듬을 이용한 광대역 음성 부호화기용 LSF 계수 양자화 장치 및 방법에 관한 것으로, LSF 계수 벡터를 양자화하여 양자화 대상 후보 벡터를 산출하고, 상기 후보 벡터를 상기 LSF 계수 벡터의 예측된 LSF 벡터를 참조하여 트렐리스 부호 양자화(TCQ)하여 상기 LSF 계수 벡터의 예측 양자화 최종 벡터를 산출하는 예측 구조 양자화부, 상기 LSF 계수 벡터를 양자화하여 양자화 대상 후보 벡터를 산출하고, 상기 후보 벡터를 트렐리스 부호 양자화(TCQ)하여 상기 LSF 계수 벡터의 비예측 양자화 최종 벡터를 산출하는 비예측 구조 양자화부 및 상기 예측 양자화 최종 벡터와 비예측 양자화 최종 벡터 중 상기 LSF 계수 벡터와의 차가 작은 것을 상기 LSF 계수 벡터의 최종 양자화 벡터로 결정하는 스위칭부로 구성되어, 안전한 양자화와 에러 전달 현상을 줄여 할당되는 비트와 SD 성능를 개선시킬 수 있는 효율적인 LSF 양자화기를 제공한다.
양자화기, BCTCQ 알고리즘, 예측 벡터, 비예측 벡터-
公开(公告)号：KR1020080102027A
公开(公告)日：2008-11-24
申请号：KR1020070048302
申请日：2007-05-17
Applicant: 한국전자통신연구원 , 한양대학교 산학협력단 , 광운대학교 산학협력단 , 연세대학교 산학협력단
CPC classification number: H03M7/40 , G10L19/0017
Abstract: A lossless coding/decoding apparatus and method of audio signal are provided to improve the performance of an audio encoder by compressing audio signal into a bit string having a less bit number at the same coding distortion. A lossless encoding apparatus comprises a second encoder module(210) including a first encoder(230) directly encoding a first symbol and a plurality of second encoders(220) converting the first symbol into a second symbol and encoding the second symbol; a first selector(250) comparing the performance of the second encoders and the first encoder and outputting an encoding mode according to the comparison result; and a second selector(260) encoding the first symbol corresponding to the encoding mode and outputting the final bit string.
Abstract translation: 提供音频信号的无损编码/解码装置和方法，以通过将音频信号压缩成具有相同编码失真的位数较少的位串来提高音频编码器的性能。 无损编码装置包括第二编码器模块（210），其包括直接编码第一符号的第一编码器（230）和将第一符号转换为第二符号并对第二符号进行编码的多个第二编码器（220） 比较第二编码器和第一编码器的性能并根据比较结果输出编码模式的第一选择器（250） 以及编码对应于编码模式的第一符号并输出最终位串的第二选择器（260）。
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公开(公告)号：KR100212447B1
公开(公告)日：1999-08-02
申请号：KR1019960056395
申请日：1996-11-22
Applicant: 한국전자통신연구원
IPC: G06F11/00
CPC classification number: G06F11/1471
Abstract: The present invention relates to a recovery function from a system failure such as a power failure and a media failure such as a breakage of disk at a transaction processing system of a plurality of user environment, and is applied in carrying out a checkpoint to reduce an amount of work required for failure recovery, and particularly to a fuzzy checkpoint method which is a method of enhancing a system performance by not interrupting other transactions ever for a moment at the time of checkpoint. According to the present invention, the occurrance of dangling transaction is prevented by deleting the corresponding transaction entry from the transaction table during performing of the redo step, for the transactions terminated between the point of time when the checkpoint start log record of the last completed checkpoint is logged and the point of time when the checkpoint end log record is logged.
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