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51. 发明申请

US20080122668A1 Scalable audio data arithmetic decoding method, medium, and apparatus, and method, medium, and apparatus truncating audio data bitstream 有权
标题翻译：可缩放音频数据算术解码方法，媒体和装置以及截取音频数据比特流的方法，媒体和装置
公开(公告)号：US20080122668A1
公开(公告)日：2008-05-29
申请号：US12000671
申请日：2007-12-14
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： H03M7/00
CPC分类号： G10L19/0017 , G10L19/24 , G10L21/038
摘要： A scalable audio data arithmetic decoding method, medium, and apparatus, and a method, medium, and apparatus truncating an audio data bitstream. The arithmetic decoding method of decoding a scalable arithmetic coded symbol may include arithmetic decoding of a symbol by using the symbol and a probability value for the symbol desired to be decoded, and determining whether or not to continue decoding by checking an ambiguity indicating whether or not decoding of the symbol to be decoded is completed. According to a method, medium, and apparatus of the present invention, data to which scalability is applied when arithmetic coding is performed in MPEG-4 scalable lossless audio coding can be efficiently decoded. Even when a bitstream is truncated, a decoding termination point can be known such that additional decoding of the truncated part can be performed.
摘要翻译：可扩展音频数据算术解码方法，介质和装置，以及截取音频数据比特流的方法，介质和装置。对可缩放算术编码符号进行解码的算术解码方法可以包括通过使用符号和期望解码的符号的概率值对符号进行算术解码，并且通过检查指示是否进行歧义来确定是否继续解码要解码的符号的解码完成。根据本发明的方法，介质和装置，可以有效地解码在MPEG-4可伸缩无损音频编码中执行算术编码时应用可扩展性的数据。即使当比特流被截断时，可以知道解码终止点，使得可以执行对截断部分的附加解码。

52. 发明申请

US20070223709A1 Method, medium, and system generating a stereo signal 有权
标题翻译：方法，介质和系统产生立体声信号
公开(公告)号：US20070223709A1
公开(公告)日：2007-09-27
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 , H04B1/00
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。

53. 发明申请

US20070115153A1 METHOD AND DEVICE TO PROVIDE ARITHMETIC DECODING OF SCALABLE BSAC AUDIO DATA 失效
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：US20070115153A1
公开(公告)日：2007-05-24
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。

54. 发明申请

US20060122825A1 Method and apparatus for transforming audio signal, method and apparatus for adaptively encoding audio signal, method and apparatus for inversely transforming audio signal, and method and apparatus for adaptively decoding audio signal 有权
标题翻译：用于变换音频信号的方法和装置，用于对音频信号进行自适应编码的方法和装置，用于对音频信号进行逆变换的方法和装置，以及用于自适应地解码音频信号的方法和装置
公开(公告)号：US20060122825A1
公开(公告)日：2006-06-08
申请号：US11295648
申请日：2005-12-07
申请人： Eunmi Oh , Junghoe Kim , Boris Kudryashov , Konstantin Osipov
发明人： Eunmi Oh , Junghoe Kim , Boris Kudryashov , Konstantin Osipov
IPC分类号： G10L19/02
CPC分类号： G10L19/022
摘要： A method and apparatus for transforming an audio signal, a method and apparatus for adaptively encoding an audio signal, a method and apparatus for inversely transforming an audio signal, and a method and apparatus for adaptively decoding an audio signal. The method of transforming an audio signal includes determining a transform unit into which the audio signal in a time domain is to be transformed into an audio signal in a frequency domain, and transforming the audio signal into an audio signal in the frequency domain according to the determined transform units using a window coefficient other than 0. Accordingly, it is possible to minimize distortion of the audio signal when encoding the audio signal even at a high bit rate while increasing efficiency of compression.
摘要翻译：用于变换音频信号的方法和装置，用于对音频信号进行自适应编码的方法和装置，用于对音频信号进行逆变换的方法和装置，以及用于对音频信号进行自适应解码的方法和装置。变换音频信号的方法包括：确定将时域中的音频信号变换为频域的音频信号的变换单元，并根据所述频域将音频信号变换为频域中的音频信号使用除了0以外的窗系数的确定的变换单元。因此，即使以高比特率对音频信号进行编码也可以最小化音频信号的失真，同时提高压缩效率。

55. 发明申请

US20060004566A1 Low-bitrate encoding/decoding method and system 审中-公开
标题翻译：低比特率编码/解码方法和系统
公开(公告)号：US20060004566A1
公开(公告)日：2006-01-05
申请号：US11165569
申请日：2005-06-24
申请人： Eunmi Oh , Junghoe Kim , Sangwook Kim , Andrew Egorov , Anton Porov , Konstantin Osipov , Boris Kudryashov
发明人： Eunmi Oh , Junghoe Kim , Sangwook Kim , Andrew Egorov , Anton Porov , Konstantin Osipov , Boris Kudryashov
IPC分类号： G10L21/00
CPC分类号： G10L19/0204 , G10L19/0017 , G10L19/032
摘要： A low-bitrate encoding system includes: a time-frequency transform unit transforming an input time-domain audio signal into a frequency-domain audio signal; a frequency component processor unit decimating frequency components in the frequency-domain audio signal; a psychoacoustic model unit modeling the received time-domain audio signal on the basis of human auditory characteristics, and calculating encoding bit allocation information; a quantizer unit quantizing the frequency-domain audio signal input from the frequency component processor unit to have a bitrate based on the encoding bit allocation information input from the psychoacoustic model unit; and a lossless encoder unit encoding the quantized audio signal losslessly, and outputting the encoded audio signal in a bitstream format. Using the low-bitrate encoding system, it is possible to effectively compress data at a low bitrate, and thus to provide a high quality audio signal.
摘要翻译：低比特率编码系统包括：时间 - 频率变换单元，将输入的时域音频信号变换为频域音频信号; 频率分量处理器单元抽取频域音频信号中的频率分量; 心理声学模型单元，基于人类听觉特征对所接收的时域音频信号进行建模，以及计算编码比特分配信息; 量化器单元，根据从心理声学模型单元输入的编码比特分配信息量化从频率分量处理器单元输入的频域音频信号以具有比特率; 以及无损地编码量化音频信号的无损编码器单元，并以比特流格式输出编码音频信号。使用低比特率编码系统，可以以低比特率有效地压缩数据，从而提供高质量的音频信号。

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