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1. 发明授权

US09934789B2 Method, medium, and apparatus with scalable channel decoding 有权
公开(公告)号：US09934789B2
公开(公告)日：2018-04-03
申请号：US11652031
申请日：2007-01-11
申请人： Junghoe Kim , Eunmi Oh , Kihyun Choo , Miao Lei
发明人： Junghoe Kim , Eunmi Oh , Kihyun Choo , Miao Lei
IPC分类号： H04R5/00 , G10L19/24 , G10L19/008
CPC分类号： G10L19/24 , G10L19/008
摘要： A method, medium, and apparatus with scalable channel decoding. The method includes recognizing the configuration of channels or speakers, calculating the respective number of same path decoding levels for each multi-channel signal using the recognized configuration of the channels or speakers, and performing decoding and up-mixing according to the calculated respective number of decoding levels.

2. 发明授权

US08595000B2 Method and apparatus to search fixed codebook and method and apparatus to encode/decode a speech signal using the method and apparatus to search fixed codebook 失效
标题翻译：使用该方法和装置搜索固定码本的方法和装置，用于搜索固定码本和方法和装置来对语音信号进行编码/解码
公开(公告)号：US08595000B2
公开(公告)日：2013-11-26
申请号：US11709255
申请日：2007-02-22
申请人： Kangeun Lee , Eunmi Oh , Hosang Sung , Changyong Son , Kihyun Choo , Junghoe Kim
发明人： Kangeun Lee , Eunmi Oh , Hosang Sung , Changyong Son , Kihyun Choo , Junghoe Kim
IPC分类号： G10L19/00 , G10L19/12 , G10L21/00
CPC分类号： G10L19/12 , G10L19/24 , G10L2019/0013
摘要： A method and an apparatus to encode and decode a speech signal using a code excited linear prediction (CELP) algorithm. In order to reduce a bit rate without degrading performance in an enhancement layer based on CELP, each of a fixed codebook of a core layer and a fixed codebook of the enhancement layer is divided into a plurality of spaces. The spaces of the fixed codebook of the enhancement layer excludes a space corresponding to a least distorted space determined from among the spaces of the fixed codebook of the core layer are searched.
摘要翻译：一种使用码激励线性预测（CELP）算法对语音信号进行编码和解码的方法和装置。为了在基于CELP的增强层中降低比特率而不降低性能的情况下，核心层的固定码本和增强层的固定码本中的每一个被分成多个空格。增强层的固定码本的空间不包括与核心层的固定码本的空格中确定的最小失真空间相对应的空间。

3. 发明授权

US07495586B2 Method and device to provide arithmetic decoding of scalable BSAC audio data 有权
标题翻译：提供可缩放BSAC音频数据的算术解码的方法和装置
公开(公告)号：US07495586B2
公开(公告)日：2009-02-24
申请号：US11957861
申请日：2007-12-17
申请人： Kihyun Choo , Junghoe Kim , Eunmi Oh , Changyong Son
发明人： Kihyun Choo , Junghoe Kim , Eunmi Oh , Changyong Son
IPC分类号： H03M7/38
CPC分类号： H03M7/30 , G10L19/24
摘要： A method and device to arithmetically decode scalable bit sliced arithmetic coding (BSAC) audio data are provided. The arithmetic decoding method includes checking whether all side information of a last layer of the BSAC audio data is to be decoded, and performing BSAC smart decoding and terminating the decoding when all the side information of the last layer is not decoded, and terminating decoding of the last layer when all the side information of the last layer is decoded. The BSAC smart decoding includes checking whether a symbol to be decoded is determined regardless of data read after the truncated bitstream, continuing the decoding when decoding can be performed regardless of data read after the truncated bitstream, and determining that ambiguity occurs and terminating the decoding, when the symbol to be decoded is determined dependent on the data read after the truncated bitstream. Accordingly, byte level scalability can be provided in a MPEG-4 BSAC decoder. Even though the bitstream is truncated, the decoding termination time can be known, and additional decoding with respect to a truncated portion of the bitstream can be performed.
摘要翻译：提供了一种用于算术解码可缩放位分片算术编码（BSAC）音频数据的方法和装置。算术解码方法包括检查BSAC音频数据的最后一层的所有侧信息是否被解码，并且当最后一层的所有侧信息未解码时，执行BSAC智能解码并终止解码，并终止解码当最后一层的所有侧面信息被解码时的最后一层。 BSAC智能解码包括检查是否确定要解码的符号，而不管截断比特流之后读取的数据如何，当可以执行解码时继续解码，而不管截断的比特流之后读取的数据如何，并且确定该模糊性并终止解码，当要解码的符号取决于在截断比特流之后读取的数据时确定。因此，可以在MPEG-4 BSAC解码器中提供字节级可扩展性。即使比特流被截断，解码终止时间可以是已知的，并且可以执行关于比特流的截断部分的附加解码。

4. 发明申请

US20080095276A1 METHOD AND DEVICE TO PROVIDE ARITHMETIC DECODING OF SCALABLE BSAC AUDIO DATA 有权
标题翻译：提供可扩展的BSAC音频数据的算术解码的方法和设备
公开(公告)号：US20080095276A1
公开(公告)日：2008-04-24
申请号：US11957861
申请日：2007-12-17
申请人： Kihyun CHOO , Junghoe Kim , Eunmi Oh , Changyong Son
发明人： Kihyun CHOO , Junghoe Kim , Eunmi Oh , Changyong Son
IPC分类号： G06F17/00 , H03D1/00
CPC分类号： H03M7/30 , G10L19/24
摘要： A method and device to arithmetically decode scalable bit sliced arithmetic coding (BSAC) audio data are provided. The arithmetic decoding method includes checking whether all side information of a last layer of the BSAC audio data is to be decoded, and performing BSAC smart decoding and terminating the decoding when all the side information of the last layer is not decoded, and terminating decoding of the last layer when all the side information of the last layer is decoded. The BSAC smart decoding includes checking whether a symbol to be decoded is determined regardless of data read after the truncated bitstream, continuing the decoding when decoding can be performed regardless of data read after the truncated bitstream, and determining that ambiguity occurs and terminating the decoding, when the symbol to be decoded is determined dependent on the data read after the truncated bitstream. Accordingly, byte level scalability can be provided in a MPEG-4 BSAC decoder. Even though the bitstream is truncated, the decoding termination time can be known, and additional decoding with respect to a truncated portion of the bitstream can be performed.
摘要翻译：提供了一种用于算术解码可缩放位分片算术编码（BSAC）音频数据的方法和装置。算术解码方法包括检查BSAC音频数据的最后一层的所有侧信息是否被解码，并且当最后一层的所有侧信息未解码时，执行BSAC智能解码并终止解码，并终止解码当最后一层的所有侧面信息被解码时的最后一层。 BSAC智能解码包括检查是否确定要解码的符号，而不管截断的比特流之后读取的数据如何，当可以执行解码时继续解码，而不管截断的比特流之后读取的数据如何，并且确定该模糊性并终止解码，当要解码的符号取决于在截断比特流之后读取的数据时确定。因此，可以在MPEG-4 BSAC解码器中提供字节级可扩展性。即使比特流被截断，解码终止时间可以是已知的，并且可以执行关于比特流的截断部分的附加解码。

5. 发明授权

US07330136B2 Method and device to provide arithmetic decoding of scalable BSAC audio data 失效
标题翻译：提供可缩放BSAC音频数据的算术解码的方法和装置
公开(公告)号：US07330136B2
公开(公告)日：2008-02-12
申请号：US11538501
申请日：2006-10-04
申请人： Kihyun Choo , Junghoe Kim , Eunmi Oh , Changyong Son
发明人： Kihyun Choo , Junghoe Kim , Eunmi Oh , Changyong Son
IPC分类号： H03M7/34
CPC分类号： H03M7/30 , G10L19/24
摘要： A method and device to arithmetically decode scalable bit sliced arithmetic coding (BSAC) audio data are provided. The arithmetic decoding method includes checking whether all side information of a last layer of the BSAC audio data is to be decoded, and performing BSAC smart decoding and terminating the decoding when all the side information of the last layer is not decoded, and terminating decoding of the last layer when all the side information of the last layer is decoded. The BSAC smart decoding includes checking whether a symbol to be decoded is determined regardless of data read after the truncated bitstream, continuing the decoding when decoding can be performed regardless of data read after the truncated bitstream, and determining that ambiguity occurs and terminating the decoding, when the symbol to be decoded is determined dependent on the data read after the truncated bitstream. Accordingly, byte level scalability can be provided in a MPEG-4 BSAC decoder. Even though the bitstream is truncated, the decoding termination time can be known, and additional decoding with respect to a truncated portion of the bitstream can be performed.
摘要翻译：提供了一种用于算术解码可缩放位分片算术编码（BSAC）音频数据的方法和装置。算术解码方法包括检查BSAC音频数据的最后一层的所有侧信息是否被解码，并且当最后一层的所有侧信息未解码时，执行BSAC智能解码并终止解码，并终止解码当最后一层的所有侧面信息被解码时的最后一层。 BSAC智能解码包括检查是否确定要解码的符号，而不管截断的比特流之后读取的数据如何，当可以执行解码时继续解码，而不管截断的比特流之后读取的数据如何，并且确定该模糊性并终止解码，当要解码的符号取决于在截断比特流之后读取的数据时确定。因此，可以在MPEG-4 BSAC解码器中提供字节级可扩展性。即使比特流被截断，解码终止时间可以是已知的，并且可以执行关于比特流的截断部分的附加解码。

6. 发明授权

US09369164B2 Method, medium, and system decoding and encoding a multi-channel signal 有权
标题翻译：方法，中，系统解码和编码多通道信号
公开(公告)号：US09369164B2
公开(公告)日：2016-06-14
申请号：US11651633
申请日：2007-01-10
申请人： Junghoe Kim , Eunmi Oh , Kihyun Choo , Miao Lei
发明人： Junghoe Kim , Eunmi Oh , Kihyun Choo , Miao Lei
IPC分类号： H04R5/00 , H04B1/16 , G10L19/008 , H04S3/02 , H04S3/00
CPC分类号： H04S5/005 , G10L19/008 , H04B1/1646 , H04S3/00 , H04S3/02 , H04S2420/01
摘要： A method, medium, and system decoding and/or encoding multiple channels. Accordingly, down-mixed multiple channels can be decoded/up-mixed to a left channel and a right channel during a first stage, thereby enabling a high quality sound output even in scalable channel decoding.
摘要翻译：方法，介质和系统解码和/或编码多个通道。因此，在第一阶段中，下混合的多个通道可以被解码/上混合到左声道和右声道，从而即使在可扩展的声道解码中也能实现高质量的声音输出。

7. 发明授权

US08214203B2 Method and apparatus for recovering line spectrum pair parameter and speech decoding apparatus using same 有权
标题翻译：用于恢复线谱对参数和使用其的语音解码装置的方法和装置
公开(公告)号：US08214203B2
公开(公告)日：2012-07-03
申请号：US12659943
申请日：2010-03-25
申请人： Hosang Sung , Seungho Choi , Kihyun Choo
发明人： Hosang Sung , Seungho Choi , Kihyun Choo
IPC分类号： G10L11/00 , G10L19/14 , G10L13/00 , G10L13/06 , G10L19/00
CPC分类号： G10L19/005 , G10L19/07
摘要： A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame. The method and apparatus can improve the quality of a recovered speech signal, be applied to a variety of technologies, and provide a method of recovering an LSP parameter for development of an algorithm for speech decoding.
摘要翻译：提供了一种用于在语音解码期间发生帧丢失时恢复频谱区的线谱对（LSP）参数的方法和装置，以及采用该频谱对参数的语音解码装置。在语音解码中恢复LSP参数的方法包括：如果确定接收到的语音分组具有已擦除的帧，则将已擦除帧的先前好帧（PGF）的LSP参数或PGF的LSP参数和将擦除的帧的下一个良好帧（NGF）进入频谱区域并获得PGF和NGF的PGF或频谱包络的频谱包络; 使用PGF的频谱包络或PGF和NGF的频谱包络来恢复被擦除的帧的频谱包络; 以及将所述已擦除帧的所恢复的频谱包络转换为所述已擦除帧的LSP参数。该方法和装置可以提高恢复的语音信号的质量，应用于各种技术，并提供一种恢复用于语音解码算法开发的LSP参数的方法。

8. 发明申请

US20070106502A1 ADAPTIVE TIME/FREQUENCY-BASED AUDIO ENCODING AND DECODING APPARATUSES AND METHODS 有权
标题翻译：自适应时间/频率的音频编码和解码设备和方法
公开(公告)号：US20070106502A1
公开(公告)日：2007-05-10
申请号：US11535164
申请日：2006-09-26
申请人： Junghoe Kim , Eunmi Oh , Changyong Son , Kihyun Choo
发明人： Junghoe Kim , Eunmi Oh , Changyong Son , Kihyun Choo
IPC分类号： G10L11/04
CPC分类号： G10L19/12 , G10L19/02 , G10L19/20
摘要： Adaptive time/frequency-based audio encoding and decoding apparatuses and methods. The encoding apparatus includes a transformation & mode determination unit to divide an input audio signal into a plurality of frequency-domain signals and to select a time-based encoding mode or a frequency-based encoding mode for each respective frequency-domain signal, an encoding unit to encode each frequency-domain signal in the respective encoding mode, and a bitstream output unit to output encoded data, division information, and encoding mode information for each respective frequency-domain signal. In the apparatuses and methods, acoustic characteristics and a voicing model are simultaneously applied to a frame, which is an audio compression processing unit. As a result, a compression method effective for both music and voice can be produced, and the compression method can be used for mobile terminals that require audio compression at a low bit rate.
摘要翻译：自适应时间/频率音频编码和解码装置和方法。编码装置包括：变换和模式确定单元，用于将输入音频信号分成多个频域信号，并且为每个相应的频域信号选择基于时间的编码模式或基于频率的编码模式，编码单元，用于对各个编码模式中的每个频域信号进行编码，以及比特流输出单元，用于输出每个各个频域信号的编码数据，分割信息和编码模式信息。在装置和方法中，声音特性和语音模型被同时应用于作为音频压缩处理单元的帧。结果，可以产生对音乐和语音有效的压缩方法，并且压缩方法可以用于需要低比特率的音频压缩的移动终端。

9. 发明授权

US09087511B2 Method, medium, and system for generating a stereo signal 有权
标题翻译：用于产生立体声信号的方法，媒体和系统
公开(公告)号：US09087511B2
公开(公告)日：2015-07-21
申请号：US11707989
申请日：2007-02-20
申请人： Junghoe Kim , Eunmi Oh , Kihyun Choo , Miao Lei
发明人： Junghoe Kim , Eunmi Oh , Kihyun Choo , Miao Lei
IPC分类号： H04R5/00 , G10L19/008 , H04S1/00 , H04S3/00 , H04S7/00
CPC分类号： H04N13/161 , G10L19/008 , H04S1/007 , H04S3/008 , H04S7/308 , H04S2420/03
摘要： Surround audio decoding for selectively generating an audio signal from a multi-channel signal. In the surround audio decoding, a down-mixed signal, e.g., as down-mixed by an encoding terminal, is selectively up-mixed to a stereo signal or a multi-channel signal, by generating spatial information for generating the stereo signal, using spatial information for up-mixing the down-mixed signal to the multi-channel signal.
摘要翻译：环绕音频解码，用于从多声道信号选择性地产生音频信号。在环绕音频解码中，通过生成用于产生立体声信号的空间信息，例如通过编码终端下混合的下混合信号被选择性地上混为立体声信号或多声道信号，使用用于将下混合信号上混合到多声道信号的空间信息。

10. 发明申请

US20100191523A1 Method and apparatus for recovering line spectrum pair parameter and speech decoding apparatus using same 有权
标题翻译：用于恢复线谱对参数和使用其的语音解码装置的方法和装置
公开(公告)号：US20100191523A1
公开(公告)日：2010-07-29
申请号：US12659943
申请日：2010-03-25
申请人： Hosang Sung , Seungho Choi , Kihyun Choo
发明人： Hosang Sung , Seungho Choi , Kihyun Choo
IPC分类号： G10L11/04
CPC分类号： G10L19/005 , G10L19/07
摘要： A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame. The method and apparatus can improve the quality of a recovered speech signal, be applied to a variety of technologies, and provide a method of recovering an LSP parameter for development of an algorithm for speech decoding.
摘要翻译：提供了一种用于在语音解码期间发生帧丢失时恢复频谱区的线谱对（LSP）参数的方法和装置，以及采用该频谱对参数的语音解码装置。在语音解码中恢复LSP参数的方法包括：如果确定接收到的语音分组具有已擦除的帧，则将已擦除帧的先前好帧（PGF）的LSP参数或PGF的LSP参数和将擦除的帧的下一个良好帧（NGF）进入频谱区域并获得PGF和NGF的PGF或频谱包络的频谱包络; 使用PGF的频谱包络或PGF和NGF的频谱包络来恢复被擦除帧的频谱包络; 以及将所述已擦除帧的所恢复的频谱包络转换为所述已擦除帧的LSP参数。该方法和装置可以提高恢复的语音信号的质量，应用于各种技术，并提供一种恢复用于语音解码算法开发的LSP参数的方法。

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